Cisco IOS Voice Command Reference

Cisco IOS Voice Command Reference

Cisco IOS Voice Command Reference - S commands

Last Modified: 2017-04-14

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C O N T E N T S

C H A P T E R 1

sccp through service-type call-check 1

sccp

3

sccp blf-speed-dial retry-interval

5

sccp ccm

7

sccp ccm group

10

sccp codec mask

12

sccp ip precedence

14

sccp local

16

sccp plar

18

sccp switchback timeout guard

19

scenario-cause

21

sdspfarm tag

23

sdspfarm transcode sessions

25

sdspfarm units

27

secondary

29

security

31

security acl

34

security izct

36

security mode

38

sequence-numbers

40

server (auto-config application)

42

server (presence)

43

server (RLM)

45

server absent reject

47

server flow-control

49

server registration-port

52

server routing

54

server trigger arq

55

Cisco IOS Voice Command Reference - S commands iii

Contents

C H A P T E R 2

server trigger brq

59

server trigger drq

62

server trigger irr

66

server trigger lcf

70

server trigger lrj

74

server trigger lrq

78

server trigger rai

82

server trigger rrq

86

server trigger urq

90

service

94

service dsapp

96

service-flow primary upstream

102

service-map

103

service-relationship

104

service-type call-check

106

session through sgcp tse payload

107

session

109

session group

111

session protocol (dial peer)

113

session protocol (Voice over Frame Relay)

115

session protocol aal2

117

session protocol multicast

118

session refresh

119

session start

120

session target (MMoIP dial peer)

122

session target (POTS dial peer)

126

session target (VoATM dial peer)

128

session target (VoFR dial peer)

132

session target (VoIP dial peer)

135

session target

141

session transport

142

session transport (H.323 voice-service)

144

session transport (SIP)

146

session-set

148

iv

Cisco IOS Voice Command Reference - S commands

Contents

C H A P T E R 3

session-timeout

150

set

151

set http client cache stale

153

set pstn-cause

154

set sip-status

157

settle-call

160

settlement

162

settlement roam-pattern

164

sgcp

166

sgcp call-agent

168

sgcp graceful-shutdown

170

sgcp max-waiting-delay

172

sgcp modem passthru

174

sgcp quarantine-buffer disable

176

sgcp request retries

178

sgcp request timeout

180

sgcp restart

182

sgcp retransmit timer

184

sgcp timer

186

sgcp tse payload

188

source filter

190

show aal2 profile through show call filter match-list

191

show aal2 profile

193

show atm video-voice address

196

show auto-config

198

show backhaul-session-manager group

200

show backhaul-session-manager session

203

show backhaul-session-manager set

206

show call accounting-template voice

208

show call active fax

212

show call active media

223

show call active total-calls

231

show call active video

232

show call active voice

243

Cisco IOS Voice Command Reference - S commands v

Contents

C H A P T E R 4

show call application app-level

266

show call application gateway-level

270

show call application interface

273

show call application services registry

277

show call application session-level

279

show call application sessions

291

show call application voice

295

show call fallback cache

301

show call fallback config

303

show call fallback stats

305

show call filter components

307

show call filter match-list

309

show call history fax through show debug condition 311

show call history fax

313

show call history media

325

show call history video

334

show call history video record

336

show call history voice

337

show call language voice

344

show call leg

346

show call media forking

351

show callmon

352

show call prompt-mem-usage

354

show call resource voice stats

358

show call resource voice threshold

360

show call rsvp-sync conf

362

show call rsvp-sync stats

364

show call spike status

366

show call threshold

368

show call treatment

371

show call-router routes

374

show call-router status

376

show ccm-manager

379

show cdapi

390

vi

Cisco IOS Voice Command Reference - S commands

Contents

C H A P T E R 5

show ces clock-select

392

show connect

393

show controllers rs366

396

show controllers timeslots

398

show controllers voice

400

show crm

403

show csm

406

show csm call

415

show cube debug category codes

417

show cube status

418

show debug condition

420

show dial-peer through show gatekeeper zone prefix 423

show dial-peer

425

show dial-peer video

426

show dial-peer voice

428

show dialplan dialpeer

439

show dialplan incall

446

show dialplan incall uri

449

show dialplan in-carrier

456

show dialplan in-trunk-group-label

458

show dialplan number

460

show dialplan uri

463

show dn-numbers

466

show dspfarm

468

show dspfarm profile

475

show dsp-group

479

show echo-cancel

481

show event-manager consumers

483

show frame-relay vofr

485

show gatekeeper calls

488

show gatekeeper circuits

491

show gatekeeper cluster

493

show gatekeeper endpoint circuits

494

show gatekeeper endpoints

497

Cisco IOS Voice Command Reference - S commands vii

Contents

C H A P T E R 6

show gatekeeper gw-type-prefix

501

show gatekeeper performance statistics

504

show gatekeeper servers

508

show gatekeeper status

511

show gatekeeper status cluster

513

show gatekeeper zone cluster

514

show gatekeeper zone prefix

515

show gatekeeper zone status

517

show gateway through show modem relay statistics

519

show gateway

521

show h323 calls preserved

523

show h323 gateway

525

show h323 gateway prefixes

533

show http client cache

535

show http client cache

539

show http client cookie

542

show http client history

544

show http client secure status

545

show http client statistics

547

show interface dspfarm

551

show interfaces cable-modem

557

show iua as

562

show iua asp

565

show media resource status

568

show mediacard

570

show mgcp

573

show mgcp connection

583

show mgcp endpoint

588

show mgcp nas

591

show mgcp profile

595

show mgcp srtp

601

show mgcp statistics

604

show modem relay statistics

609

viii

Cisco IOS Voice Command Reference - S commands

Contents

C H A P T E R 7

show mrcp client session active through show sip dhcp 615

show monitor event-trace voip ccsip (EXEC)

617

show mrcp client session active

632

show mrcp client session history

635

show mrcp client statistics hostname

639

show mwi relay clients

641

show nextport

643

show nextport vpd

649

show num-exp

651

show piafs status

653

show pots csm

655

show pots status

656

show pots volume

660

show presence global

661

show presence subscription

664

show proxy h323 calls

668

show proxy h323 detail-call

669

show proxy h323 status

674

show raw

675

show rawmsg

678

show rlm group statistics

680

show rlm group status

683

show rlm group timer

685

show rpms-proc counters

688

show running-config dial-peer

690

show rtpspi

692

show rtsp client session

694

show rudpv0 failures

697

show rudpv0 statistics

699

show rudpv1

701

show sccp

704

show sccp ccm group

712

show sccp connections details

715

show sccp connections internal

716

Cisco IOS Voice Command Reference - S commands ix

Contents

C H A P T E R 8

show sccp connections rsvp

717

show sccp connections summary

719

show sccp server statistics

721

show sdspfarm

722

show settlement

728

show sgcp connection

731

show sgcp endpoint

733

show sgcp statistics

734

show shared-line

736

show sip dhcp

739

show sip service through show trunk hdlc

741

show sip service

743

show sip-ua calls

744

show sip-ua connections

755

show sip-ua map

760

show sip-ua min-se

763

show sip-ua mwi

765

show sip-ua register status

767

show sip-ua retry

769

show sip-ua service

772

show sip-ua srtp

774

show sip-ua statistics

776

show sip-ua status

785

show sip-ua status refer-ood

789

show sip-ua timers

791

show spe voice

794

show ss7 mtp1 channel-id

797

show ss7 mtp1 links

800

show ss7 mtp2 ccb

803

show ss7 mtp2 state

806

show ss7 mtp2 stats

810

show ss7 mtp2 timer

814

show ss7 mtp2 variant

816

show ss7 sm session

819

x

Cisco IOS Voice Command Reference - S commands

Contents

C H A P T E R 9

show ss7 sm set

821

show ss7 sm stats

825

show stcapp buffer-history

827

show stcapp device

829

show stcapp feature codes

836

show stcapp statistics

840

show subscription

842

show subscription local

849

show tbct

857

show tdm mapping

858

show tgrep neighbors

860

show translation-rule

862

show trunk group

865

show trunk hdlc

869

show vdev through show voice statistics memory-usage

871

show vdev

873

show vfc

876

show vfc cap-list

877

show vfc default-file

878

show vfc directory

879

show vfc version

881

show video call summary

885

show voice accounting method

886

show voice accounting response pending

889

show voice busyout

890

show voice cable-status

892

show voice call

894

show voice call rate

905

show voice cause-code

907

show voice class called-number

909

show voice class called-number-pool

911

show voice class e164-pattern-map

913

show voice class phone-proxy

915

show voice class resource-group

917

Cisco IOS Voice Command Reference - S commands xi

Contents

C H A P T E R 1 0

show voice class uri

919

show voice connectivity summary

922

show voice data

923

show voice dnis-map

926

show voice dsmp stream

929

show voice dsp

936

show voice dsp channel

943

show voice dsp crash-dump

946

show voice dsp summary

948

show voice eddri prefix

951

show voice enum-match-table

953

show voice hpi capture

955

show voice iec description

957

show voice lmr

959

show voice pcm capture

965

show voice permanent-call

966

show voice port

971

show voice source-group

983

show voice statistics csr interval accounting

986

show voice statistics csr interval aggregation

989

show voice statistics csr since-reset accounting

995

show voice statistics csr since-reset aggregation-level

998

show voice statistics csr since-reset all

1004

show voice statistics iec

1009

show voice statistics interval-tag

1012

show voice statistics memory-usage

1014

show voice trace through shutdown (voice-port)

1017

show voice trace

1019

show voice translation-profile

1020

show voice translation-rule

1022

show voice trunk-conditioning signaling

1024

show voice trunk-conditioning supervisory

1027

Show voice vrf

1030

show voice vtsp

1031

xii

Cisco IOS Voice Command Reference - S commands

Contents

C H A P T E R 1 1

show voip debug version

1033

show voip fpi call-rate

1034

show voip fpi calls

1035

show voip fpi rtts

1041

show voip fpi stats

1042

show voip htsp

1044

show voip recmsp session

1046

show voip rtp connections

1048

show voip rtp forking

1052

show voip trunk group

1054

show vrm active_calls

1055

show vrm vdevices

1058

show vsp

1069

show wsapi

1074

show xcsp port

1078

show xcsp slot

1081

shut

1083

shutdown (Annex G neighbor)

1084

shutdown (Annex G)

1085

shutdown (dial-peer)

1086

shutdown (DSP Farm profile)

1087

shutdown (gatekeeper)

1088

shutdown (gateway)

1090

shutdown (mediacard)

1091

shutdown (auto-config application)

1092

shutdown (RLM)

1093

shutdown (settlement)

1095

shutdown (voice-port)

1097

signal through srv version 1099

signal

1101

signal did

1106

signal keepalive

1107

signal pattern

1109

signal sequence oos

1113

Cisco IOS Voice Command Reference - S commands xiii

Contents

C H A P T E R 1 2

signal timing idle suppress-voice

1116

signal timing oos

1119

signal timing oos restart

1122

signal timing oos slave-standby

1124

signal timing oos suppress-all

1126

signal timing oos suppress-voice

1128

signal timing oos timeout

1130

signaling forward

1132

signaling forward (dial peer)

1137

signal-type

1143

silent-discard untrusted

1146

silent-fax

1147

sip

1149

sip-header

1151

sip-header SIP-StatusLine

1152

sip-server

1153

sip-ua

1156

snmp enable peer-trap dscp-profile

1159

snmp enable peer-trap poor-qov

1160

snmp-server enable traps voice (DSCP profile)

1161

soft-offhook

1163

source-address (uc-wsapi)

1165

source carrier-id

1166

source filter

1167

source trunk-group-label

1168

speed dial

1169

srtp (dial peer)

1172

srtp (voice)

1174

srtp-auth

1176

srtp-crypto

1178

srtp negotiate

1180

srv version

1182

ss7 mtp2-variant through switchover method

1185

ss7 mtp2-variant

1188

xiv

Cisco IOS Voice Command Reference - S commands

Contents

ss7 mtp2-variant bellcore

1196

ss7 mtp2-variant itu

1199

ss7 mtp2-variant ntt

1202

ss7 mtp2-variant ttc

1205

ss7 mtp2-variant itu-white

1208

ss7 session

1211

ss7 session cumack_t

1214

ss7 session kp_t

1216

ss7 session m_cumack

1218

ss7 session m_outseq

1220

ss7 session m_rcvnum

1222

ss7 session m_retrans

1224

ss7 session retrans_t

1226

ss7 set

1228

ss7 set failover-timer

1230

station-id name

1232

station-id number

1234

stats

1236

stcapp

1238

stcapp call-control mode

1240

stcapp feature callback

1242

stcapp ccm-group

1243

stcapp feature access-code

1245

stcapp feature callback

1247

stcapp feature speed-dial

1248

stcapp register capability

1250

stcapp security mode

1252

stcapp security trustpoint

1254

stcapp supplementary-services

1256

stcapp timer

1257

stun

1258

stun flowdata agent-id

1259

stun flowdata catlife

1261

stun flowdata keepalive

1262

stun flowdata shared-secret

1264

Cisco IOS Voice Command Reference - S commands xv

Contents

stun usage firewall-traversal flowdata

1266

stun usage ice lite

1267

subaddress

1268

subcell-mux

1269

subscription asnl session history

1270

subscription maximum

1272

supervisory answer dualtone

1274

supervisory custom-cptone

1276

supervisory disconnect

1278

supervisory disconnect anytone

1279

supervisory disconnect dualtone

1281

supervisory disconnect dualtone voice-class

1283

supervisory disconnect lcfo

1285

supervisory dualtone-detect-params

1286

supervisory sit us

1288

supplementary-service h225-notify cid-update (dal peer)

1291

supplementary-service h225-notify cid-update (voice-service)

1293

supplementary-service h450.2 (dial peer)

1295

supplementary-service h450.2 (voice-service)

1297

supplementary-service h450.3 (dial peer)

1299

supplementary-service h450.3 (voice-service)

1301

supplementary-service h450.7

1303

supplementary-service h450.12 (dial peer)

1305

supplementary-service h450.12 (voice-service)

1307

supplementary-service media-renegotiate

1309

supplementary-service qsig call-forward

1310

supplementary-service sip

1312

supported language

1315

suppress

1317

survivability single-register

1318

suspend-resume (SIP)

1319

switchback interval

1320

switchback method

1322

switchover method

1324

xvi

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check

sccp, page 3

sccp blf-speed-dial retry-interval, page 5

sccp ccm, page 7

sccp ccm group, page 10

sccp codec mask, page 12

sccp ip precedence, page 14

sccp local, page 16

sccp plar, page 18

sccp switchback timeout guard, page 19

scenario-cause, page 21

sdspfarm tag, page 23

sdspfarm transcode sessions, page 25

sdspfarm units, page 27

secondary, page 29

security, page 31

security acl, page 34

security izct, page 36

security mode, page 38

sequence-numbers, page 40

server (auto-config application), page 42

server (presence), page 43

server (RLM), page 45

server absent reject, page 47

server flow-control, page 49

Cisco IOS Voice Command Reference - S commands

1

server registration-port, page 52

server routing, page 54

server trigger arq, page 55

server trigger brq, page 59

server trigger drq, page 62

server trigger irr, page 66

server trigger lcf, page 70

server trigger lrj, page 74

server trigger lrq, page 78

server trigger rai, page 82

server trigger rrq, page 86

server trigger urq, page 90

service, page 94

service dsapp, page 96

service-flow primary upstream, page 102

service-map, page 103

service-relationship, page 104

service-type call-check, page 106

sccp through service-type call-check

2

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sccp

sccp

To enable the Skinny Client Control Protocol (SCCP) protocol and its related applications (transcoding and conferencing), use the sccp command in global configuration mode. To disable the protocol, use the no form of this command.

sccp no sccp

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled

Command Modes

Global configuration (config)

Command History

Release

12.1(5)YH

12.2(13)T

Modification

This command was introduced on the Cisco VG200.

This command was implemented on the Cisco 2600 series, Cisco 3620,

Cisco 3640, Cisco 3660, and Cisco 3700 series.

Usage Guidelines

The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms

(NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources.

SCCP and its related applications (transcoding and conferencing) become enabled only if digital-signal-processor (DSP) resources for these applications are configured, DSP-farm service is enabled, and the Cisco CallManager registration process is completed.

The no form of this command disables SCCP and its applications by unregistering from the active Cisco

CallManager, dropping existing connections, and freeing allocated resources.

Examples

The following example sets related values and then enables SCCP:

Router(config)#

sccp ccm 10.10.10.1 priority 1

Router(config)#

sccp local fastEthernet 0/0

Router(config)#

sccp switchback timeout guard 180

Router(config)#

sccp ip precedence 5

Router(config)#

sccp

Router(config)#

end

Cisco IOS Voice Command Reference - S commands

3

sccp

Related Commands

Command dspfarm (DSP farm) show dspfarm show sccp sccp through service-type call-check

Description

Enables DSP-farm service.

Displays summary information about DSP resources.

Displays the SCCP configuration information and current status.

4

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sccp blf-speed-dial retry-interval

sccp blf-speed-dial retry-interval

To set the retry timeout for Busy Lamp Field (BLF) notification for speed-dial numbers on SCCP phones registered to an external Cisco Unified CME router, use the sccp blf-speed-dial retry-intervalcommand in presence configuration mode. To reset to the default, use the no form of this command.

sccp blf-speed-dial retry-interval seconds limit number

no sccp blf-speed-dial retry-interval

Syntax Description

seconds

limit number

Retry timeout in seconds. Range: 60 to 3600. Default:

60.

Maximum number of retries. Range: 10 to 100.

Default: 10.

Command Default

Retry timeout is 60 seconds; retry limit is 10.

Command Modes

Presence configuration (config-presence)

Command History

Cisco IOS Release

12.4(11)XJ

12.4(15)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.4(15)T.

Usage Guidelines

This command specifies how frequently the router attempts to subscribe for the line status of an external directory number when the BLF speed-dial feature is configured on a SCCP phone. This retry mechanism is used when either the presentity does not exist or the router receives a terminated NOTIFY from the external presence server. If the subscribe request toward the external server fails after the configured number of retries, the subscribe request from the phone is rejected.

Examples

The following example shows the BLF speed-dial retry interval set to 100 seconds and the limit to 25:

Router(config)#

presence

Router(config-presence)#

sccp blf-speed-dial retry-interval 100 limit 25

Cisco IOS Voice Command Reference - S commands

5

sccp blf-speed-dial retry-interval

Related Commands

Command allow subscribe blf-speed-dial server show presence global sccp through service-type call-check

Description

Allows internal watchers to monitor external presence entities (directory numbers).

Enables BLF monitoring for a speed-dial number on a phone registered to Cisco Unified CME.

Specifies the IP address of a presence server for sending presence requests from internal watchers to external presence entities.

Displays configuration information about the presence service.

6

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sccp ccm

sccp ccm

To add a Cisco Unified Communications Manager server to the list of available servers and set various parameters--including IP address or Domain Name System (DNS) name, port number, and version number--use the sccp ccm command in global configuration mode. To remove a particular server from the list, use the no form of this command.

NM-HDV or NM-HDV-FARM Voice Network Modules

sccp ccm {ipv4-address| ipv6-address| dns} priority priority [port port-number] [version version-number]

[trustpoint label]

no sccp ccm {ipv4-address| ipv6-address| dns}

NM-HDV2 or NM-HD-1V/2V/2VE Voice Network Modules

sccp ccm {ipv4-address| ipv6-address| dns} identifier identifier-number [priority priority] [port port-number]

[version version-number] [trustpoint label]

no sccp ccm {ipv4-address| ipv6-address| dns}

Syntax Description

ipv4 -address

ipv6-address dns

identifier

identifier-number

priority

priority

port port -number

IPv4 address of the Cisco Unified Communications

Manager server.

IPv6 address of the Cisco Unified Communications

Manager server.

DNS name.

Specifies the number that identifies the Cisco Unified

Communications Manager server. The range is 1 to

65535.

Specifies the priority of this Cisco Unified

Communications Manager server relative to other connected servers. The range is 1 (highest) to 4

(lowest).

Note

This keyword is required only for NM-HDV and NM-HDV-FARM modules. Do not use this keyword if you are using the NM-HDV2 or NM-HD-1V/2V/2VE; set the priority using the associate ccm command in the

Cisco Unified Communications Manager group.

(Optional) Specifies the TCP port number. The range is 1025 to 65535. The default is 2000.

Cisco IOS Voice Command Reference - S commands

7

sccp through service-type call-check sccp ccm

version version -number

trustpoint

label

(Optional) Cisco Unified Communications Manager version. Valid versions are 3.0, 3.1, 3.2, 3.3, 4.0, 4.1,

5.0.1, 6.0, and 7.0+. There is no default value.

(Optional) Specifies the trustpoint for Cisco Unified

Communications Manager certificate.

Cisco Unified Communications Manager trustpoint label.

Command Default

The default port number is 2000.

Command Modes

Global configuration (config)

Command History

Release

12.1(5)YH

12.3(8)T

12.4(11)XW

12.4(20)T

12.4(22)T

15.0(1)M

Modification

This command was introduced.

This command was modified. The identifier keyword and additional values for

Cisco Unified Communications Manager versions were added.

This command was modified. The 6.0 keyword was added to the list of version values.

This command was integrated into Cisco IOS Release 12.4(20)T.

This command was modified. Support for IPv6 was added. The version keyword and version-numberargumentwere changed from being optional to being required, and the 7.0+ keyword was added.

This command was modified in a release earlier than Cisco IOS Release

15.0(1)M. The trustpointkeyword and the label argument were added.

Usage Guidelines

You can configure up to four Cisco Unified Communications Manager servers--a primary and up to three backups--to support digital signal processor (DSP) farm services. To add the Cisco Unified Communications

Manager server to a Cisco Unified Communications Manager group, use the associate ccm command.

IPv6 support is provided for registration with Cisco Unified CM version 7.0 and later.

To enable Ad Hoc or Meet-Me hardware conferencing in Cisco Unified CME, you must first set the version keyword to 4.0 or a later version.

Beginning with Cisco IOS Release 12.4(22)T users manually configuring the sccp ccmcommand must provide the version. Existing router configurations are not impacted because automatic upgrade and downgrade are supported.

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Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sccp ccm

Examples

The following example shows how to add the Cisco Unified Communications Manager server with IP address

10.0.0.0 to the list of available servers:

Router(config)#

sccp ccm 10.0.0.0 identifier 3 port 1025 version 4.0

The following example shows how to add the Cisco Unified CallManager server whose IPv6 address is

2001:DB8:C18:1::102:

Router(config)#

sccp ccm 2001:DB8:C18:1::102 identifier 2 version 7.0

Related Commands

Command associate ccm sccp sccp ccm group sccp local show sccp

Description

Associates a Cisco Unified Communications Manager server with a Cisco Unified Communications Manager group and establishes its priority within the group.

Enables SCCP and its associated transcoding and conferencing applications.

Creates a Cisco Unified Communications Manager group and enters SCCP Cisco Unified

Communications Manager configuration mode.

Selects the local interface that SCCP applications use to register with Cisco Unified Communications

Manager.

Displays SCCP configuration information and current status.

Cisco IOS Voice Command Reference - S commands

9

sccp through service-type call-check sccp ccm group

sccp ccm group

To create a Cisco Unified Communications Manager group and enter SCCP Cisco CallManager configuration mode, use the sccp ccm group command in global configuration mode. To remove a particular Cisco Unified

Communications Manager group, use the no form of this command.

sccp ccm group group-number

no sccp ccm group group-number

Syntax Description

group-number

Number that identifies the Cisco Unified

Communications Manager group. Range is 1 to 50.

Command Default

No groups are defined, so all servers are configured individually.

Command Modes

Global configuration (config)

Command History

Release

12.3(8)T

12.4(22)T

15.0(1)M

Modification

This command was introduced.

This command was modified. Support for IPv6 was added.

This command was modified. The group number range was increased to 50.

Usage Guidelines

Use this command to group Cisco Unified Communications Manager servers that are defined using the sccp

ccm command. You can associate designated DSP farm profiles using the associate profile command so that the DSP services are controlled by the Cisco Unified Communications Manager servers in the group.

Examples

The following example enters SCCP Cisco CallManager configuration mode and associates Cisco Unified

Communications Manager 25 with Cisco Unified Communications Manager group 10:

Router(config)#

sccp ccm group 10

Router(config-sccp-ccm)#

associate ccm 25 priority 2

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sccp through service-type call-check

Related Commands

Command associate ccm associate profile bind interface connect interval connect retries sccp ccm sccp ccm group

Description

Associates a Cisco Unified Communications Manager server with a Cisco Unified Communications Manager group and establishes its priority within the group.

Associates a DSP farm profile with a Cisco Unified

Communications Manager group.

Binds an interface with a Cisco Unified

Communications Manager group.

Specifies the amount of time that a DSP farm profile waits before attempting to connect to a Cisco Unified

Communications Manager when the current Cisco

Unified Communications Manager fails to connect.

Specifies the number of times that a DSP farm attempts to connect to a Cisco Unified

Communications Manager when the current Cisco

Unified Communications Manager connections fails.

Adds a Cisco Unified Communications Manager server to the list of available servers.

Cisco IOS Voice Command Reference - S commands

11

sccp through service-type call-check sccp codec mask

sccp codec mask

To mask a codec type so that it is not used by Cisco CallManager, use the sccp codec maskcommand in global configuration mode. To unmask a codec, use the no form of this command.

sccp codec codec mask

no sccp codec codec mask

Syntax Description

codec

Codec to mask. Values are the following:

g711alaw

g711ulaw

g729abr8

g729ar8

g729br8

g729r8

Command Default

No codecs are masked.

Command Modes

Global configuration (config)

Command History

Release

12.1(5)YH4

12.2(13)T

12.4(11)XJ2

12.4(15)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(13)T.

The gsmefrand gsmfrkeywords were removed as configurable codec options for all platforms with the exception of the gsmfr codec on the Cisco AS5400 and AS5350 with MSAv6 dsps.

This command was integrated into Cisco IOS Release 12.4(15)T.

Usage Guidelines

This command prevents the voice gateway from reporting codec types that are masked so that Cisco

CallManager only selects codec types that are supported by the endpoints.

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sccp through service-type call-check sccp codec mask

Note

You must enable this command before Skinny Client Control Protocol (SCCP) is enabled. If the sccp

codec mask command is used when SCCP is active, you must disable the SCCP using the no sccp command and then re-enable sccp for the sccp codec mask command to take effect.

Examples

The following example shows how to mask codec type G.711 ulaw and G.729r8: sccp codec g711ulaw mask sccp codec g729r8 mask

Related Commands

Command sccp sccp ccm sccp local show sccp

Description

Enables SCCP and related applications.

Adds a Cisco CallManager server to the list of available servers and sets various parameters.

Selects the local interface that SCCP applications use to register with Cisco CallManager.

Displays SCCP configuration information and current status.

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13

sccp through service-type call-check sccp ip precedence

sccp ip precedence

To set the IP precedence value to be used by Skinny Client Control Protocol (SCCP), use the sccp ip precedence command in global configuration mode. To reset to the default, use the no form of this command.

sccp ip precedence value

no sccp ip precedence

Syntax Description

value

IP precedence value. Range is from 1 (lowest) to 7

(highest).

Command Default

5

Command Modes

Global configuration (config)

Command History

Release

12.1(5)YH

12.2(13)T

Modification

This command was introduced on the Cisco VG200.

This command was implemented on the Cisco 2600 series, Cisco 3620,

Cisco 3640, Cisco 3660, and Cisco 3700 series.

Usage Guidelines

The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms

(NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources.

Examples

The following example sets IP precedence to the highest possible value:

Router#

sccp ip precedence 1

Related Commands

Command dspfarm (DSP farm) sccp show sccp

Description

Enables DSP-farm service.

Enables SCCP and its associated transcoding and conferencing applications.

Displays the SCCP configuration information and current status.

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sccp through service-type call-check sccp ip precedence

Cisco IOS Voice Command Reference - S commands

15

sccp through service-type call-check sccp local

sccp local

To select the local interface that Skinny Client Control Protocol (SCCP) applications (transcoding and conferencing) use to register with Cisco CallManager, use the sccp local command in global configuration mode. To deselect the interface, use the no form of this command.

sccp local interface-type interface-number [port port-number]

no sccp local interface-type interface-number

Syntax Description

interface -type

interface-number

port port-number

Interface type that the SCCP application uses to register with Cisco CallManager. The type can be an interface address or a virtual-interface address such as Ethernet.

Interface number that the SCCP application uses to register with Cisco CallManager.

(Optional) Port number used by the selected interface.

Range is 1025 to 65535. Default is 2000.

Command Default

No default behavior or values

Command Modes

Global configuration (config)

Command History

Release

12.1(5)YH

12.2(13)T

12.3(14)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(13)T.

The port keyword and port-number argument were added.

Usage Guidelines

The router must be equipped with one or more voice network modules that provide DSP resources.

Note

If the default port is used by another application, the SCCP application fails to register to Cisco

CallManager. Use the port keyword with the port-numberargument to specify a different port for SCCP to use for registering with Cisco CallManager.

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sccp through service-type call-check sccp local

Examples

The following example selects a Fast Ethernet interface for SCCP applications to use to register with Cisco

CallManager: sccp local FastEthernet 0/0

Related Commands

Command dsp services dspfarm sccp show sccp

Description

Enables DSP-farm services.

Enables SCCP and its associated transcoding and conferencing applications.

Displays the SCCP configuration information and current status.

Cisco IOS Voice Command Reference - S commands

17

sccp through service-type call-check sccp plar

sccp plar

To enter SCCP PLAR configuration mode, use the sccp plar command in global configuration mode. To disable private line automatic ringdown (PLAR) on all ports, use the no form of this command.

sccp plar no sccp plar

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled (PLAR is not enabled on any port).

Command Modes

Global configuration (config)

Command History

Release

12.4(6)T

Modification

This command was introduced.

Usage Guidelines

This command is used for enabling PLAR features on analog FXS endpoints that use Skinny Client Control

Protocol (SCCP) for call control. Use the voiceport command to enable a specific analog voice port for PLAR.

Examples

The following example sets PLAR on voice ports 2/0, 2/1, and 2/3:

Router(config)#

sccp plar

Router(config-sccp-plar)#

voiceport 2/0 dial 3660 digit 1234 wait-connect 500 interval 200

Router(config-sccp-plar)#

voiceport 2/1 dial 3264 digit 678,,,9*0,,#123 interval 100

Router(config-sccp-plar)#

voiceport 2/3 dial 3478 digit 34567 wait-connect 500

Related Commands

Command dial peer voice voiceport

Description

Enters dial-peer configuration mode and defines a dial peer.

Enables a PLAR connection for an analog phone.

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Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sccp switchback timeout guard

sccp switchback timeout guard

To set the Skinny Client Control Protocol (SCCP) switchback guard timer, use the sccp switchback timeout

guard command in global configuration mode. To reset to the default, use the no form of this command.

sccp switchback timeout guard seconds

no sccp switchback timeout guard

Syntax Description

seconds

Guard timer value, in seconds. Range is from 180 to

7200. Default is 1200.

Command Default

1200 seconds

Command Modes

Global configuration (config)

Command History

Release

12.1(5)YH

12.2(13)T

Modification

This command was introduced on the Cisco VG200.

This command was implemented on the Cisco 2600 series, Cisco 3620,

Cisco 3640, Cisco 3660, and Cisco 3700 series.

Usage Guidelines

The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms

(NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources.

You can use the guard timer value for the switchback algorithm that follows the Graceful Timer method.

Examples

The following example sets the switchback guard timer value to 180 seconds (3 minutes):

Router#

sccp switchback timeout guard 180

Related Commands

Command dspfarm (DSP farm) sccp

Description

Enables DSP-farm service.

Enables SCCP and its associated transcoding and conferencing applications.

Cisco IOS Voice Command Reference - S commands

19

sccp switchback timeout guard

Command show sccp sccp through service-type call-check

Description

Displays the SCCP configuration information and current status.

20

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check scenario-cause

scenario-cause

To configure new Q.850 call-disconnect cause codes for use if an H.323 call fails, use the scenario-cause command in H.323-voice-service configuration mode. To revert to the defaults, use the no form of this command.

scenario-cause {arj-default| timeout {arq| t301| t303| t310} code-id}

no scenario-cause {arj-default| timeout {arq| t301| t303| t310}}

Syntax Description

arj-default code-id

timeout arq code-id

timeout t301 code-id

timeout t303 code-id

timeout t310 code-id

Q.850 call-disconnect cause code for use if a call fails for reasons that are assigned to the Admission Reject

(ARJ) default cause code. Range: 1 to 127.

Q.850 call-disconnect cause code for use if the H.323

gatekeeper Automatic Repeat Request (ARQ) timer expires. Range: 1 to 127.

Q.850 call-disconnect cause code for use when the

H.225 alerting (T301) timer expires. Range: 1 to 127.

Q.850 call-disconnect cause code for use when the

H.225 setup (T303) timer expires. Range: 1 to 127.

Q.850 call-disconnect cause code for use when the

H.225 call-proceeding (T310) timer expires. Range:

1 to 127.

Command Default

No mapping occurs.

Command Modes

H.323 voice-service configuration (conf-serv-h323)

Command History

Release

12.4(9)T

Modification

This command was introduced.

Usage Guidelines

Use this command to configure new Q.850 call-disconnect cause codes for use if an H.323 voice call fails during setup.

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21

sccp through service-type call-check scenario-cause

Examples

The following example causes a gateway to send the default ARJ cause code of 24 rather than the previous default of 63 when a call fails for reasons that are associated with the ARJ default cause code:

Router(config)#

voice service voip

Router(conf-voi-serv)#

h323

Router(conf-serv-h323)#

scenario-cause arj-default 24

Related Commands

Command h225 timeout call-proceeding map q850-cause q850-cause

Description

Sets the call-proceeding (T310, or call-setup to call-disconnect) disconnect timer.

Maps a Q.850 call-disconnect cause code to a tone.

Maps a Q.850 call-disconnect cause code to a different Q.850 call-disconnect cause code.

22

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sdspfarm tag

sdspfarm tag

To permit a digital-signal-processor (DSP) farm to be to registered to Cisco Unified CME and associate it with the MAC address of a Skinny Client Control Protocol (SCCP) interface, use the sdspfarm tag command in telephony-service configuration mode. To delete a tag generated by the sdspfarm tagcommand, use the

no form of this command.

sdspfarm tag number device-name

no sdspfarm tag number device-name

Syntax Description

number device-name

Numeric name for a DSP farm. Number from 1 to 10.

Word describing the device, such as the MAC address, of the SCCP client interface that is preceded by the Message Transfer Part (MTP).

Command Default

DSP farm is not created.

Command Modes

Telephony-service configuration (config-telephony)

Command History

Cisco IOS Release

12.3(11)T

15.1(4)M

Cisco Product

Cisco CME 3.2

Cisco CME 8.6

Modification

This command was introduced.

This command was modified. The maximum number used to tag a DSP farm was increased to

10.

Usage Guidelines

DSP farm profiles are sets of DSP resources used for conferencing and transcoding only. DSP farms do not include voice termination resources. Use the show interfacecommand to find the MAC address of the SCCP client interface.

Examples

The following example declares tag 1 as the MAC address of mac000a.8aea.ca80. The show interface command is used to obtain the MAC address.

Router#

show interface FastEthernet 0/0

.

.

.

FastEthernet0/0 is up, line protocol is up

Hardware is AmdFE, address is 000a.8aea.ca80 (bia 000a.8aea.ca80)

Cisco IOS Voice Command Reference - S commands

23

sccp through service-type call-check sdspfarm tag

.

.

.

Router(config)#

telephony-service

Router(config-telephony)#

sdspfarm tag 1 mac000a.8aea.ca80

Related Commands

Command sdspfarm transcode sdspfarm units

Description

Specifies the maximum number of transcoding sessions allowed per Cisco CME router.

Specifies the maximum number of DSP farms that are allowed to be registered to the SCCP server.

24

Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check sdspfarm transcode sessions

sdspfarm transcode sessions

To specify the maximum number of transcoding sessions allowed per Cisco CallManager Express (Cisco

CME) router, use the sdspfarm transcode sessionscommand in telephony-service configuration mode. To return to the default transcode session of 0, use the no form of this command.

sdspfarm transcode sessions number

no sdspfarm transcode sessions number

Syntax Description

number

Declares the number of DSP farm sessions. Valid values are numbers from 1 to 128.

Command Default

The default is 0.

Command Modes

Telephony-service configuration (config-telephony)

Command History

Cisco IOS Release

12.3(11)T

Cisco Product

Cisco CME 3.2

Modification

This command was introduced.

Usage Guidelines

The transcoding is allowed between G.711 and G.729. A session consists of two transcode streams. To configure this information, you must know how many digital-signal-processor (DSP) farms are configured on the network module (NM) farms in your Cisco CME router. DSP farms are sets of DSP resources used for conferencing and transcoding only. DSP farms do not include voice termination resources. To learn how many

DSP farms have been configured on your Cisco CME router, use the show sdspfarm command.

Examples

The following example sets the maximum number of transcoding sessions allowed on the Cisco CME router to 20:

Router(config)#

telephony-service

Router(config-telephony)#

sdspfarm transcode sessions 20

Related Commands

Command sdspfarm tag

Description

Declares a DSP farm and associates it with an SCCP client interface’s MAC address.

Cisco IOS Voice Command Reference - S commands

25

sdspfarm transcode sessions

Command sdspfarm unit show sdspfarm sccp through service-type call-check

Description

Specifies the maximum number of DSP farms that are allowed to be registered to the SCCP server.

Displays the status of the configured DSP farms and transcoding streams.

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sccp through service-type call-check sdspfarm units

sdspfarm units

To specify the maximum number of digital-signal-processor (DSP) farm profiles that are allowed to be registered to the Skinny Client Control Protocol (SCCP) server, use the sdspfarm units command in telephony-service configuration mode. To set the number of DSP farm profiles to the default value of 0, use the no form of this command.

sdspfarm units number

no sdspfarm units number

Syntax Description

number

Number of DSP farms. Valid values are numbers from

0 to 10.

Command Default

The default number is 0.

Command Modes

Telephony-service configuration (config-telephony)

Command History

Cisco IOS Release

12.3(11)T

15.1(4)M

Cisco Product

Cisco CME 3.2

Cisco CME 8.6

Modification

This command was introduced.

This command was modified. The command increased support for the maximum number of DSP farms to 10.

Usage Guidelines

DSP farm profiles are sets of DSP resources used for conferencing and transcoding only. DSP farm profiles do not include voice termination resources.

Examples

The following example configures a Cisco CME router to register one DSP farm:

Router(config)#

telephony-service

Router(config-telephony)#

sdspfarm units 1

Related Commands

Command sdspfarm tag

Description

Declares a DSP farm and associates it with the MAC address of an SCCP client interface.

Cisco IOS Voice Command Reference - S commands

27

sdspfarm units

Command sdspfarm transcode sccp through service-type call-check

Description

Specifies the maximum number of transcoding sessions allowed per Cisco CME router.

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Cisco IOS Voice Command Reference - S commands

sccp through service-type call-check secondary

secondary

To set the backup location for storing call detail records (CDRs) if the primary location becomes unavailable, use the secondary command in gateway accounting file configuration mode. To reset to the default, use the

no form of this command.

secondary {ftp path/filename username username password password | ifs device:filename}

no secondary {ftp| ifs}

Syntax Description

ftp path /filename

ifs device : filename

username username

password password

Name and location of the backup file on an external

FTP server. Filename is limited to 25 characters.

Name and location of the backup file in flash memory or other internal file system on this router. Values depend on storage devices available on the router, for example flash or slot0. Filename is limited to 25 characters.

User ID for authentication.

Password user enters for authentication.

Command Default

Call records are saved to flash:cdr.

Command Modes

Gateway accounting file configuration (config-gw-accounting-file)

Command History

Release

12.4(15)XY

12.4(20)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

This command defines the backup location where accounting records are sent if the file transfer to the primary device fails. The file accounting process retries the primary device, defined with the primary command, up to the number of times defined by the maximum retry-count command before automatically switching over to the secondary device.

The secondary device is attempted only after the primary device fails after the defined number of retries. If the secondary device also fails, the system logs an error and the file accounting process stops.

To manually switch back to the primary device when it becomes available, use the file-acct reset command.

The system does not automatically switch back to the primary device.

Cisco IOS Voice Command Reference - S commands

29

sccp through service-type call-check secondary

Examples

A syslog warning message is generated if flash becomes full.

The filename you assign is appended with the gateway hostname and time stamp at the time the file is created to make the filename unique. For example, if you specify the filename cdrtest1 on a router with the hostname cme-2821, a file is created with the name cdrtest1.cme-2821.2007_10_28T22_21_41.000, where

2007_10_28T22_21_41.000 is the time that the file was created.

Limit the filename you assign with this command to 25 characters, otherwise it could be truncated when the accounting file is created because the full filename, including the appended hostname and timestamp, is limited to 63 characters.

The following example shows the backup location of the accounting file is set to flash:cdrtest2: gw-accounting file primary ftp server1/cdrtest1 username bob password temp secondary ifs flash:cdrtest2 maximum buffer-size 25 maximum retry-count 3 maximum fileclose-timer 720 cdr-format compact

Related Commands

Command file-acct reset maximum retry-count primary

Description

Manually switches back to the primary device for file accounting.

Sets the maximum number of times the router attempts to connect to the primary file device before switching to the secondary device.

Sets the primary location for storing the CDRs generated for file accounting.

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sccp through service-type call-check security

security

To enable authentication and authorization on a gatekeeper, use the security command in gatekeeper configuration mode. To disable security, use the no form of this command.

security {any| h323-id| e164} {password default password| password separator character}

no security {any| h323-id| e164} {password default password| password separator character}

Syntax Description any

h323 -id

e164

password default password

password separator character

Uses the first alias of an incoming registration, admission, and status (RAS) protocol registration, regardless of its type, to identify the user to

RADIUS/TACACS+.

Uses the first H.323 ID type alias to identify the user to RADIUS/TACACS+.

Uses the first E.164 address type alias to identify the user to RADIUS/TACACS+.

Default password that the gatekeeper associates with endpoints when authenticating them with an authentication server. The password must be identical to the password on the authentication server.

Character that endpoints use to separate the H.323-ID from the piggybacked password in the registration.

Specifying this character allows each endpoint to supply a user-specific password. The separator character and password are stripped from the string before it is treated as an H.323-ID alias to be registered.

Note that passwords may only be piggybacked in the

H.323-ID, not the E.164 address, because the E.164

address allows a limited set of mostly numeric characters. If the endpoint does not wish to register an H.323-ID, it can still supply an H.323-ID consisting of just the separator character and password. This H.323-ID consisting of just the separator character and password are understood to be a password mechanism and no H.323-ID is registered.

Command Default

No default

Cisco IOS Voice Command Reference - S commands

31

sccp through service-type call-check security

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

11.3(2)NA

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series.

Usage Guidelines

Use this command to enable identification of registered aliases by RADIUS/TACACS+. If the alias does not exist in RADIUS/TACACS+, the endpoint is not allowed to register.

A RADIUS/TACACS+ server and encryption key must have been configured in Cisco IOS software for security to work.

Only the first alias of the proper type is identified. If no alias of the proper type is found, the registration is rejected.

This command does not allow you to define the password mechanism unless the security type (h323-id or

e164 or any) has been defined. Although the no security password command undefines the password mechanism, it leaves the security type unchanged, so security is still enabled. However, the no security command disables security entirely, including removing any existing password definitions.

Examples

The following example enables identification of registrations using the first H.323 ID found in any registration: security h323id

The following example enables security, authenticating all users by using their H.323-IDs and a password of qwerty2x: security h323-id security password qwerty2x

The next example enables security, authenticating all users by using their H.323-IDs and the password entered by the user in the H.323-ID alias he or she registers: security h323-id security password separator !

If a user registers with an H.323-ID of joe!024aqx, the gatekeeper authenticates user joe with password 024aqx, and if that is successful, registers the user with the H.323-ID of joe. If the exclamation point is not found, the user is authenticated with the default password, or a null password if no default has been configured.

The following example enables security, authenticating all users by using their E.164 IDs and the password entered by the user in the H.323-ID alias he or she registers: security e164 security password separator !

If a user registers with an E.164 address of 5551212 and an H.323-ID of !hs8473q6, the gatekeeper authenticates user 5551212 and password hs8473q6. Because the H.323-ID string supplied by the user begins with the separator character, no H.323-ID is registered, and the user is known only by the E.164 address.

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Related Commands

Command accounting (gatekeeper) radius-server host

radius -server key

security

Description

Enables the accounting security feature on the gatekeeper.

Specifies a RADIUS server host.

Sets the authentication and encryption key for all

RADIUS communications between the router and the

RADIUS daemon.

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33

sccp through service-type call-check security acl

security acl

To configure access-list based filtering on the gatekeeper, use the security aclcommand in gatekeeper configuration mode. To disable, use the no form of this command.

security acl {answerarq| lrq} access-list-number

no security acl {answerarq| lrq}

Syntax Description

answerarq

lrq

access-list-number

Filters incoming answer admission requests

(AnswerARQ) using IP access-lists.

Filters incoming location requests (LRQs) using IP access-list.

Number of an access list that was configured using the access-list command. This is a decimal number from 1 to 99 or from 1300 to 1999. Only standard IP access lists numbered 1 through 99 are supported for the Tokenless Call Authorization feature.

Command Default

No default behavior or values.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.3(5)

Modification

This command was introduced.

Usage Guidelines

The security acl command configures the gatekeeper to use IP access lists for security. Use this command in conjunction with the access-list command to configure access-list based AnswerARQ and LRQ requests filtering on a gatekeeper. The gatekeeper will process only those requestswhich have been sent by sources that are permitted access by the specified IP access list. Requests sent by sources which have been denied by the specified IP access lists, will be rejected.

Examples

The following example shows how to configure a gatekeeper to use a previously configured IP access list with an IP access list number of 30 for call authorization:

Router(config-gk)#

security acl answerarq 30

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The following example shows how to configure a gatekeeper to use a previously configured IP access list with an IP access list number of 20 for LRQ filtering:

Router(config-gk)#

security acl lrq 20

Related Commands

Command

access -list

Description

Configures the access list mechanism for filtering frames by protocol type or vendor code.

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35

sccp through service-type call-check security izct

security izct

To configure the gatekeeper to include the destination E.164 alias in the IZC token hash, use the security izct command in gatekeeper configuration mode. To not include destination E.16 alias in IZC token hash, use the

no form of this command.

security izct password password [hash {dest-alias| src-alias| dest-csa| src-csa| dest-epid| src-epid}]

no security izct password [hash {dest-alias| src-alias| dest-csa| src-csa| dest-epid| src-epid}]

Syntax Description

password password

hash dest-alias src-alias dest-csa src-csa dest-epid src-epid

Specifies the password that the gatekeeper associates with endpoints when authenticating them with an authentication server. The password must be identical to the password on the authentication server.

Specifies the options to be used in hash generation.

Specifies that the destination alias be included in hash generation.

Specifies that the source alias be included in hash generation.

Specifies that the destination csa be included in hash generation.

Specifies that the source alias be included in hash generation.

Specifies that the destination epid be included in hash generation.

Specifies that the source epid be included in hash generation.

Command Default

Destination E.16 alias are not included in IZC token hash.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.3(5)

Modification

This command was introduced.

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Release

12.4(15)XZ

12.4(20)T

Modification

The dest-alias, src-alias, dest-csa, src-csa, dest-epid,and

src-epidkeywords were added.

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

Configure the security izct command on the gatekeeper that generates the InterZone Clear Token (IZCT) hash to prevent rogue endpoints from sending an ARQ message with one called number and then changing the called number when they send the SETUP message to the terminating endpoint. When this command is configured, modification of the called number after the IZCT hash is generated by the trunking gateway will not be allowed. The IZCT token generated is valid only for 30 seconds and the IZCT hash token generated by terminating gatekeeper (TGK) can be used for multiple calls.

The call is rejected if any intermediate entity, such as a Cisco Gatekeeper Transaction Message Protocol

(GKTMP) server (on the originating gatekeeper) or the originating gateway (using number translation rules), tries to modify the called number after the token is prepared during address resolution.

• The hash keyword at originating gateway (OGW) and TGK do not need to match.

• More than one hash keyword can be configured for the security izctcommand.

The security izctcommand must be configured at OGK or TGK in order to enable the feature.

When configuring an OGW to an OGK to a TGK and to a TGW. The security izctcommand is optional at the OGK, and required at the TGK. If hash parameter is not specified at the TGK, then dest-alias (default) will be used for hash token computation.

The no version of this command the requires the keyword argument combinations as defined in the preceding command syntax table.

Examples

The following example prevents modification of the called number after the IZCT hash is generated by the trunking gateway:

Router(config-gk)#

security izct password example hash dest-alias

Related Commands

Command accounting (gatekeeper) radius-server host radius-server key

Description

Enables the accounting security feature on the gatekeeper.

Specifies a RADIUS server host.

Sets the authentication and encryption key for all

RADIUS communications between the router and the

RADIUS daemon.

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sccp through service-type call-check security mode

security mode

To set the security mode for a specific dial peer using Skinny Client Control Protocol (SCCP) Telephony

Control Application (STCAPP) services in a secure Cisco Unified CME network, use the security mode command in dial peer configuration mode. To return to the default, use the no form of this command.

security mode {authenticated| none| encrypted| system}

no security mode

Syntax Description authenticated none encrypted system

Sets the security mode to authenticated and enables

SCCP signaling between the voice gateway and Cisco

Unified CME to take place through the secure TLS connection on TCP port 2443.

SCCP signaling is not secure.

Sets the security mode to encrypted and enables SCCP signaling between the voice gateway and Cisco

Unified CME to take place through Secure Real-Time

Transport Protocol (SRTP).

Enables the security mode specified at the global level by the stcapp security mode command.

Command Default

Security mode specified at the global level is enabled.

Command Modes

Dial peer configuration (config-dialpeer)

Command History

Release

12.4(11)XW1

12.4(20)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

Use this command to specify security mode on the voice gateway for Cisco Unified CME phone authentication and encryption.

Set the SCCP signaling security mode globally using the stcapp security mode command in global configuration mode. If you use both the stcapp security mode and the security modecommands, the dial-peer level command, security mode, overrides the global setting.

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Examples

The following example selects secure SCCP signaling in authenticated mode:

Router(config)#

dial-peer voice 1 pots

Router(config-dialpeer)#

security mode authenticated

The following example selects encrypted secure SCCP signaling and encryption through SRTP:

Router(config)#

dial-peer voice 2 pots

Router(config-dialpeer)#

security mode encrypted

Related Commands

Command stcapp security mode

Description

Enables security for STCAPP endpoints and specifies the security mode to be used for setting up the TLS connection.

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39

sccp through service-type call-check sequence-numbers

sequence-numbers

To enable the generation of sequence numbers in each frame generated by the digital signal processor (DSP) for Voice over Frame Relay applications, use the sequence-numbers command in dial-peer configuration mode. To disable the generation of sequence numbers, use the no form of this command.

sequence-numbers no sequence-numbers

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release

12.0(3)XG

12.0(4)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was integrated into Cisco IOS Release 12.0(4)T.

Usage Guidelines

Sequence numbers on voice packets allow the digital signal processor (DSP) at the playout side to detect lost packets, duplicate packets, or out-of-sequence packets. This helps the DSP to mask out occasional drop-outs in voice transmission at the cost of one extra byte per packet. The benefit of using sequence numbers versus the cost in bandwidth of adding an extra byte to each voice packet on the Frame Relay network must be weighed to determine whether to disable this function for your application.

Another factor to consider is that this command does not affect codecs that require a sequence number, such as G.726. If you are using a codec that requires a sequence number, the DSP generates one regardless of the configuration of this command.

Examples

The following example disables generation of sequence numbers for VoFR frames for VoFR dial peer 200: dial-peer voice 200 vofr no sequence-numbers

Related Commands

Command

called -number (dial-peer)

Description

Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11

trunk connection.

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Command

codec (dial -peer)

cptone

destination -pattern

dtmf -relay (Voice over Frame Relay)

session protocol (Voice over Frame Relay) session target

signal -type

Description

Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.

Specifies a regional analog voice interface-related tone, ring, and cadence setting.

Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.

Enables the generation of FRF.11 Annex A frames for a dial peer.

Establishes a session protocol for calls between the local and remote routers via the packet network.

Specifies a network-specific address for a specified dial peer or destination gatekeeper.

Sets the signaling type to be used when connecting to a dial peer.

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sccp through service-type call-check server (auto-config application)

server (auto-config application)

To configure the IP address or name of the TFTP server for an auto-configuration application, use the server command in auto-config application configuration mode. To remove the IP address or name, use the no form of this command.

{server ip-address| domain-name [ip-address| domain-name] [ip-address| domain-name]}

no server

Syntax Description

ip-address domain-name

Specifies the IP address of the TFTP server.

Specifies the domain name of the TFTP server.

Command Default

No default behavior or values.

Command Modes

Auto-config application configuration (auto-config-app)

Command History

Release

12.3(8)XY

12.3(14)T

Modification

This command was introduced on the Communication Media Module.

This command was integrated into Cisco IOS Release 12.3(14)T.

Examples

The following example shows the server command used to configure two TFTP servers for an auto-configuration application:

Router(auto-config-app)#

server 172.18.240.45 172.18.240.55

Related Commands

Command auto-config show auto-config

Description

Enables auto-configuration or enters auto-config application configuration mode for the Skinny Client

Control Protocol (SCCP) application.

Displays the current status of auto-config applications.

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server (presence)

To specify the IP address of a presence server for sending presence requests from internal watchers to external presence entities, use the servercommand in presence configuration mode. To remove the server, use the no form of this command.

server ip-address

no server

Syntax Description

ip-address

IP address of the remote presence server.

Command Default

A remote presence server is not used.

Command Modes

Presence configuration (config-presence)

Command History

Release

12.4(11)XJ

12.4(15)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.4(15)T.

Usage Guidelines

This command specifies the IP address of a presence server that handles presence requests when the watcher and presence entity (presentity) are not collocated. The router acts as the presence server and processes all presence requests and status notifications when a watcher and presentity are both internal. If a subscription request is for an external presentity, the request is sent to the remote server specified by this command.

Examples

The following example shows a presence server with IP address 10.10.10.1:

Router(config)#

presence

Router(config-presence)#

allow subscribe

Router(config-presence)#

server 10.10.10.1

Related Commands

Command allow subscribe

Description

Allows internal watchers to monitor external presence entities (directory numbers).

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43

server (presence)

Command allow watch max-subscription show presence global show presence subscription watcher all sccp through service-type call-check

Description

Allows a directory number on a phone registered to

Cisco Unified CME to be watched in a presence service.

Sets the maximum number of concurrent watch sessions that are allowed.

Displays configuration information about the presence service.

Displays information about active presence subscriptions.

Allows external watchers to monitor internal presence entities (directory numbers).

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server (RLM)

To identify an RLM server, use the server RLM configuration command. To remove the identification, use the no form of this command

server name-tag

no server name-tag

Syntax Description

name -tag Name to identify the server configuration so that multiple entries of server configuration can be entered.

Command Default

Disabled

Command Modes

RLM configuration

Command History

Release

11.3(7)

Modification

This command was introduced.

Usage Guidelines

Each server can have multiple entries of IP addresses or aliases.

Examples

The following example identifies the RLM server and defines the associated IP addresses: rlm group 1 server r1-server link address 10.1.4.1 source Loopback1 weight 4 link address 10.1.4.2 source Loopback2 weight 3

Related Commands

Command clear interface clear rlm group interface link (RLM)

Description

Resets the hardware logic on an interface.

Clears all RLM group time stamps to zero.

Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.

Specifies the link preference.

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45

server (RLM)

Command protocol rlm port retry keepalive show rlm group statistics show rlm group status show rlm group timer shutdown (RLM) timer sccp through service-type call-check

Description

Reconfigures the port number for the basic RLM connection for the whole rlm-group.

Allows consecutive keepalive failures a certain amount of time before the link is declared down.

Displays the network latency of the RLM group.

Displays the status of the RLM group.

Displays the current RLM group timer values.

Shuts down all of the links under the RLM group.

Overwrites the default setting of timeout values.

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server absent reject

To configure the gatekeeper to reject new registrations or calls when the connection to the Gatekeeper

Transaction Message Protocol (GKTMP) server is down, use the server absent reject command in gatekeeper configuration mode. To disable, use the no form of this command.

server absent reject {arq| rrq}

no server absent reject {arq| rrq}

Syntax Description arq rrq

Reject call admission request (ARQ) messages.

Reject registration request (RRQ) messages.

Command Default

By default, registrations and calls are not rejected.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.2(11)T

Modification

This command was introduced on the Cisco 3660 and Cisco MC3810.

Usage Guidelines

This command configures the gatekeeper to reject new registrations or calls when it is unable to reach the

GKTMP server because the TCP connection between the gatekeeper and GKTMP server is down. If multiple

GKTMP servers are configured, the gatekeeper tries all of them and rejects registrations or calls only if none of the servers respond. You can also use this feature for security or service denial if a connection with the server is required to complete a registration.

Note

This command assumes that RRQ and ARQ triggers are used between the gatekeeper and GKTMP server.

Examples

The following example directs the gatekeeper to reject registrations when it cannot connect to the GKTMP server:

Router#

show gatekeeper configuration

.

.

.

h323id tet gw-type-prefix 1#* default-technology gw-type-prefix 9#* gw ipaddr 1.1.1.1 1720

Cisco IOS Voice Command Reference - S commands

47

server absent reject

.

.

.

no shutdown server absent reject rrq

sccp through service-type call-check

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server flow-control

To enable flow control on the Cisco IOS gatekeeper (GK) and reset all thresholds to default, use the server

flow-control command in gatekeeper configuration mode. To disable GK flow control, use the no form of this command.

server flow-control [onset value] [abatement value] [qcount value]

no server flow-control

Syntax Description

onset value

abatement value

qcount value

(Optional) Percentage of the server timeout value that is used to mark the server as usable or unusable.

Range is from 1 to 100. The default is 80.

(Optional) Percentage of the server timeout value that is used to mark the server as unusable or usable.

Range is from 1 to 100. The default is 50.

Note

The abatement value must be less than the onset value.

(Optional) Threshold length of the outbound queue on the GK. The queue contains messages waiting to be transmitted to the server. The TCP socket between the GK and Gatekeeper Transaction Message Protocol

(GKTMP) server queues messages if it has too many to transmit. If the count of outbound queue length on the server reaches the qcount value, the server is marked unusable. Range is from 1 to 1000. The default is 400.

Command Default

The gatekeeper will send a maximum of 1000 RRQ messages.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.2(2)XB

12.2(8)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(8)T.

Usage Guidelines

Suppose the server timeout value is 3 seconds, the onset value is 50, and the abatement value is 40. When the average response time from the server to the Gatekeeper Transaction Message Protocol (GKTMP) reaches

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sccp through service-type call-check server flow-control

Examples

1.5 seconds (the onset percentage of the server timeout value), the server is marked as unusable. During the period that the server is marked as unusable, REQUEST ALV messages are still sent to the unusable server.

When the response time is lowered to 1.2 seconds (the abatement percentage of the timeout value), the server is marked usable again, and the GKTMP resumes sending messages to the server.

When the server flow-control command is configured on its own the default is value 400. If you change one parameter using the server flow-control command, all other parameters revert to the default values. For example, if the onset is configured at 70 percent and you use the server flow-controlcommand to set the abatement level, the onset resets to the default (80 percent).

The following example uses the command with the default values:

Router#

server flow-control

The following example enables the GKTMP Interface Resiliency Enhancement feature with an onset level of

50:

Router#

server flow-control onset 50

*Mar 8 20:05:34.081: gk_srv_handle_flowcontrol: Flow control enabled

Router#

show running-config

Building configuration...

Current configuration : 1065 bytes

!

version 12.2

no service single-slot-reload-enable service timestamps debug datetime msec service timestamps log uptime no service password-encryption

!

hostname snet-3660-3

!

.

.

.

gatekeeper

!

.

.

.

end zone local snet-3660-3 cisco.com

zone remote snet-3660-2 cisco.com 209.165.200.225 1719 zone prefix snet-3660-2 408* lrq forward-queries no use-proxy snet-3660-3 default inbound-to terminal no use-proxy snet-3660-3 default outbound-from terminal no shutdown server registration-port 8000 server flow-control onset 50

The following example enables the GKTMP Interface Resiliency Enhancement feature:

Router#

show gatekeeper status

Gatekeeper State: UP

Load Balancing: DISABLED

Flow Control: ENABLED

Zone Name:

Accounting:

Endpoint Throttling:

Security: snet-3660-3

DISABLED

DISABLED

DISABLED

Maximum Remote Bandwidth:

Current Remote Bandwidth: unlimited

0 kbps

Current Remote Bandwidth (w/ Alt GKs): 0 kbps

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The following example shows the server statistics, including timeout encountered, average response time, and the server status:

Router#

show gatekeeper server

GATEKEEPER SERVERS STATUS

=========================

Gatekeeper Server listening port: 8250

Gatekeeper Server timeout value: 30 (100ms)

GateKeeper GKTMP version: 3.1

Gatekeeper-ID: Gatekeeper1

------------------------

RRQ Priority: 5

Server-ID: Server43

Server IP address: 209.165.200.254:40118

Server type: dynamically registered

Connection Status: active

Trigger Information:

Trigger unconditionally

Server Statistics:

REQUEST RRQ Sent=0

RESPONSE RRQ Received = 0

RESPONSE RCF Received = 0

RESPONSE RRJ Received = 0

Timeout encountered=0

Average response time(ms)=0

Server Usable=TRUE

Related Commands

Command timer server timeout

Description

Specifies the timeout value for a response from a back-end GKTMP server.

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sccp through service-type call-check server registration-port

server registration-port

To configure the listener port for the server to establish a connection with the gatekeeper, use the server

registration-port command in gatekeeper configuration mode. To force the gatekeeper to close the listening socket so that no more new registration takes place, use the no form of this command.

server registration-port port-number

no server registration-port port-number

Syntax Description

port -number Port number on which the gatekeeper listens for external server connections. Range is from 1 to 65535.

There is no default.

Command Default

No registration port is configured.

Note

If the gatekeeper is to communicate with network servers, a registration port must be configured on it.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced on the following platforms: Cisco 2500 series,

Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.

This command was implemented on the Cisco 3700 series.

Usage Guidelines

Use this command to configure a server registration port to poll for servers that want to establish connections with the gatekeeper.

Note

The no form of this command forces the gatekeeper on this router to close the listen socket, so it cannot accept more registrations. However, existing connections between the gatekeeper and servers are left open.

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Examples

The following example establishes a listener port for a server connection with a gatekeeper:

Router(config)#

gatekeeper

Router(config-gk)#

server registration-port 20000

Related Commands

Command server trigger show gatekeeper servers

Description

Configure static server triggers for specific RAS messages to be forwarded to a specified server.

Displays the triggers configured on the gatekeeper.

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sccp through service-type call-check server routing

server routing

To specify the type of circuit messages sent to the Gatekeeper Transaction Message Protocol (GKTMP) server, use the server routingcommand in gatekeeper configuration mode. To return to the default, use the no form of this command.

server routing {both| carrier| trunk-group}

no server routing {both| carrier| trunk-group}

Syntax Description both carrier

trunk -group

Sends both types of information in GKTMP messages.

Sends only carrier information in GKTMP messages.

This is the default.

Sends only trunk-group information in GKTMP messages.

Command Default

Carrier

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.2(11)T

Modification

This command was introduced.

Usage Guidelines

Use this command to route carrier and trunk-group messages from the gatekeeper to the GKTMP server.

The carrierkeyword sends the "I" and "J" tags in the GKTMP messages. The trunk-groupkeyword sends the "P" and "Q" tags in the GKTMP messages. The bothkeyword sends both sets of tags.

Examples

The following example enables trunk-group information to be sent in GKTMP messages from the gatekeeper:

Router(config)#

gatekeeper

Router(config-gk)#

server routing trunk-group

Related Commands

Command show gatekeeper servers

Description

Displays the triggers configured on the gatekeeper.

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server trigger arq

To configure the admission request (ARQ) trigger statically on the gatekeeper, use the server trigger

arqcommand in gatekeeper configuration mode. Submode commands are available after the server trigger

arqcommand is entered. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of this command.

server trigger arq gkid priority server-id server-ip-address server-port

no server trigger arq gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

info -only Use to indicate to the Cisco IOS gatekeeper that messages that meet the specified trigger parameters should be sent to the GKTMP server application as notifications only and that the gatekeeper should not wait for a response from the Gatekeeper Transaction

Message Protocol (GKTMP) server application.

shutdown

Use to temporarily disables a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

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55

sccp through service-type call-check server trigger arq

destination -info e164 | email-id | h323-id value Use to send ARQ RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions

e164 -- Destination is an E.164 address.

email -id-- Destination is an e-mail ID.

h323 -id-- Destination is an H.323 ID.

value -- Value against which to compare the destination address in the RAS messages. For

E.164 addresses, the following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

redirect -reason reason-number

Use to send ARQ RAS messages containing a specific redirect reason to the GKTMP server application.

reason -number-- Range is from 0 to 65535.

Currently-used values are:

• 0 -- Unknown reason.

• 1 -- Call forwarding busy or called DTE busy.

• 2 -- Call forwarded; no reply.

• 4 -- Call deflection.

• 9 -- Called DTE out of order.

• 10 -- Call forwarding by the called DTE.

• 15 -- Call forwarding unconditionally.

Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration(config-gk)

Command History

Release

12.1(1)T

Modification

This command was introduced.

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Release

12.2(11)T

Modification

This command was implemented on the Cisco 2600 series, Cisco 3600 series,

Cisco 3700 series, Cisco 7200 series, and Cisco MC3810. The irrtrigger was added.

Usage Guidelines

Use this command and its optional submode commands to configure the admission request (ARQ) static server trigger. The gatekeeper checks incoming gateway ARQ messages for the configured trigger information. If an incoming ARQ message contains the specified trigger information, the gatekeeper sends the ARQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the ARQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the ARQ messages, the gatekeeper sends all ARQ messages to the GKTMP server application.

If the gatekeeper receives an ARQ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming ARQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two ARQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

ARQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming ARQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one ARQ trigger registration message with the same priority but for different GKTMP servers, the gatekeeper retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all ARQ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger arq sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_arqtrigger)#

exit

The following example configures an ARQ trigger registration on gatekeeper "alpha", which sends to GKTMP server "Server-west" any ARQ message that contains H.323 ID "3660-gw1", e-mail ID "joe.xyz.com", or a redirect reason 1. All other ARQ messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger arq alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-arqtrigger)#

destination-info h323-id 3660-gw1

Router(config-gk-arqtrigger)#

destination-info email-id joe.xyz.com

Router(config-gk-arqtrigger)#

redirect-reason 1

Router(config-gk-arqtrigger#

exit

If the ARQ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger arq alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_arqtrigger)#

destination-info e164 1800....

Router(config-gk_arqtrigger)#

exit

Then gatekeeper "alpha" checks all incoming ARQ messages for the destination H.323 ID, e-mail ID, or redirect reason before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the ARQ message to the GKTMP server "Server-west".

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If the second gatekeeper "alpha" ARQ trigger registration had been defined with a priority 1 instead of priority

2, the second server trigger definition would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those ARQ messages that contain a destination

E.164 address that starts with 1800. All other ARQ messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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server trigger brq

To configure the bandwidth request (BRQ) trigger statically on the gatekeeper, use the server trigger

brqcommand in gatekeeper configuration mode. Submode commands are available after entering the server

trigger brqcommand. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger brq gkid priority server-id server-ip-address server-port

no server trigger brq gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

info -only Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

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redirect -reason reason-number

shutdown

Use to send BRQ RAS messages containing a specific redirect reason to the GKTMP server application.

reason-number-- Range is from 0 to 65535.

Currently used values are as follows:

• 0 -- Unknown reason.

• 1 -- Call forwarding busy or called DTE busy.

• 2 -- Call forwarded; no reply.

• 4 -- Call deflection.

• 9 -- Called DTE out of order.

• 10 -- Call forwarding by the called DTE.

• 15 -- Call forwarding unconditionally.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.2(2)XB

12.2(11)T

Modification

This command was introduced.

This the command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco

MC3810. The irrtrigger was added.

Usage Guidelines

Use this command and its optional submode commands to configure the bandwidth request (BRQ) static server trigger. The gatekeeper checks incoming gateway BRQ messages for the configured trigger information.

If an incoming BRQ message contains the specified trigger information, the gatekeeper sends the BRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the BRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the BRQ messages, the gatekeeper sends all BRQ messages to the GKTMP server application.

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Examples

If the gatekeeper receives BRQ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming BRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two BRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

BRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming BRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one BRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all BRQ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger brq sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_brqtrigger)#

exit

The following example configures BRQ trigger registration on gatekeeper "alpha", which sends to GKTMP server "Server-west" any BRQ message containing redirect reason 1 or redirect reason 2. All other BRQ messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger brq alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-brqtrigger)#

redirect-reason 1

Router(config-gk-brqtrigger)#

redirect-reason 2

Router(config-gk-brqtrigger#

exit

If the BRQ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger brq alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_brqtrigger)#

redirect-reason 10

Router(config-gk_brqtrigger)#

exit

Then gatekeeper "alpha" checks all incoming BRQ messages for redirect reasons 1 or 2 before checking for redirect reason 10. If any one of those conditions is met, the gatekeeper sends the BRQ message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" BRQ trigger registration had been defined with a priority 1 instead of priority

2, then the second server trigger definition would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those BRQ messages that contain a redirect reason

10. All other BRQ messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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server trigger drq

To configure the disengage request (DRQ) trigger statically on the gatekeeper, use the server trigger

drqcommand in gatekeeper configuration mode. Submode commands are available after entering the server

trigger drqcommand. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger drq gkid priority server-id server-ip-address server-port

no server trigger drq gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the Reliability, Availability, and Serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

info -only Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

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destination -info e164 | email-id | h323-id value Use to send automatic repeat request (ARQ) RAS messages containing a specified destination to the

GKTMP server application. Configure one of the following conditions:

e164 -- Destination is an E.164 address.

email -id-- Destination is an e-mail ID.

h323 -id-- Destination is an H.323 ID.

value-- Value against which to compare the destination address in the RAS messages. For

E.164 addresses, the following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

call info type { fax | modem | voice

shutdown

Use to send ARQ RAS messages containing a specified call info type to the GKTMP server application. The following types can be used:

fax

• modem

voice

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.2(2)XB

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco

MC3810.

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Release

12.4(4)T

Modification

The call-info-type submode command was added.

Usage Guidelines

Use this command and its optional submode commands to configure the disengage request (DRQ) static server trigger. The gatekeeper checks incoming gateway DRQ messages for the configured trigger information. If an incoming DRQ message contains the specified trigger information, the gatekeeper sends the DRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the DRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the DRQ messages, the gatekeeper sends all DRQ messages to the GKTMP server application.

If the gatekeeper receives a DRQ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming DRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two DRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

DRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming DRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one DRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeeper retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper together with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all DRQ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger drq sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_drqtrigger)#

exit

The following example configures DRQ trigger registration on gatekeeper "alpha", which sends to GKTMP server "Server-west" any DRQ message containing an H.323 ID "3660-gw1" or e-mail ID "joe.xyz.com". All other DRQ messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger drq alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-drqtrigger)#

destination-info h323-id 3660-gw1

Router(config-gk-drqtrigger)#

destination-info email-id joe.xyz.com

Router(config-gk-drqtrigger#

exit

If the DRQ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger drq alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_drqtrigger)#

destination-info e164 1800....

Router(config-gk_drqtrigger)#

exit

then gatekeeper "alpha" checks all incoming DRQ messages for the destination H.323 ID or e-mail ID before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the DRQ message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" DRQ trigger registration had been defined with a priority 1 instead of priority

2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper

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"alpha" would send to GKTMP server Server-west only those DRQ messages that contain a destination E.164

address starting with 1800. All other DRQ messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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server trigger irr

To configure the information request response (IRR) trigger statically on the gatekeeper, use the server trigger

irrcommand in gatekeeper configuration mode. Submode commands are available after entering the server

trigger irrcommand. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger irr gkid priority server-id server-ip-address server-port

no server trigger irr gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

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destination -info e164 | email-id | h323-id value Use to send IRR RAS messages containing a specified destination to the GKTMP server application.

Configure one of the following conditions:

e164 -- Destination is an E.164 address.

email -id-- Destination is an e-mail ID.

h323 -id-- Destination is an H.323 ID.

value-- Value against which to compare the destination address in the RAS messages. For

E.164 addresses, the following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

info -only

redirect -reason reason-number

shutdown

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to send IRR RAS messages containing a specific redirect reason to the GKTMP server application.

reason -number--Range is from 0 to 65535.

Currently used values are the following:

• 0 -- Unknown reason.

• 1 -- Call forwarding busy or called DTE busy.

• 2 -- Call forwarded; no reply.

• 4 -- Call deflection.

• 9 -- Called DTE out of order.

• 10 -- Call forwarding by the called DTE.

• 15 -- Call forwarding unconditionally.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

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Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series,

Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.

The irrtrigger was added.

Usage Guidelines

Use this command and its optional submode commands to configure the information request response (IRR) static server trigger. The gatekeeper checks incoming gateway IRR messages for the configured trigger information. If an incoming IRR message contains the specified trigger information, the gatekeeper sends the

IRR message to the GKTMP server application. In addition, the IRR message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the IRR messages, the gatekeeper sends all IRR messages to the

GKTMP server application.

If the gatekeeper receives an IRR trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming IRR RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two IRR trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

IRR trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming IRR messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one IRR trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all IRR messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger irr sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_irrtrigger)#

exit

The following example configures an IRR trigger registration on gatekeeper "alpha", which send to GKTMP server "Server-west" any IRR message containing an H.323 ID "3660-gw1", e-mail ID "joe.xyz.com, or a redirect reason 1. All other IRR messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger irr alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-irrtrigger)#

destination-info h323-id 3660-gw1

Router(config-gk-irrtrigger)#

destination-info email-id joe.xyz.com

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Router(config-gk-irrtrigger)#

redirect-reason 1

Router(config-gk-irrtrigger#

exit

If the IRR registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger irr alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_irrtrigger)#

destination-info e164 1800....

Router(config-gk_irrtrigger)#

exit

Then gatekeeper "alpha" checks all incoming IRR messages for the destination H.323 ID, e-mail ID, or redirect reason before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the IRR message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" IRR trigger registration had been defined with a priority 1 instead of priority

2, then the second server trigger definition would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those IRR messages that contain a destination E.164

address starting with 1800. All other IRR messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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server trigger lcf

To configure the location confirm (LCF) trigger statically on the gatekeeper, use the server trigger lcfcommand in gatekeeper configuration mode. Submode commands are available after entering the server trigger

lcfcommand. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger lcf gkid priority server-id server-ip-address server-port

no server trigger lcf gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the RAS message. These filters are optional, and you may configure any of them, one per command line.

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destination -info e164 | email-id | h323-id value Use to send LCF RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions:

e164 -- Destination is an E.164 address.

email -id-- Destination is an e-mail ID.

h323 -id-- Destination is an H.323 ID.

value-- Value against which to compare the destination address in the RAS messages. For

E.164 addresses, the following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

info -only

remote -ext-address e164 value

shutdown

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to send LCF RAS messages that contain a specified remote extension address to the GKTMP server application.

e164 --Remote extension address is an E.164

address.

value --Value against which to compare the destination address in the RAS messages. The following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

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Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series,

Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.

The irrtrigger was added.

Usage Guidelines

Use this command and its optional submode commands to configure the location confirm (LCF) static server trigger. The gatekeeper checks incoming gateway LCF messages for the configured trigger information. If an incoming LCF message contains the specified trigger information, the gatekeeper sends the LCF message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the LCF message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the LCF messages, the gatekeeper sends all LCF messages to the GKTMP server application.

If the gatekeeper receives an LCF trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming LCF RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two LCF trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

LCF trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming LCF messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one LCF trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all LCF messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger lcf sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_lcftrigger)#

exit

The following example configures an LCF trigger registration on gatekeeper "alpha", which send to GKTMP server "Server-west" any LCF message containing an H.323 ID "3660-gw1", e-mail ID joe.xyz.com, or a remote extension address starting with 1408. All other LCF messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger lcf alpha 1 Server-west 10.10.10.10 1751

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Router(config-gk-lcftrigger)#

destination-info h323-id 3660-gw1

Router(config-gk-lcftrigger)#

destination-info email-id joe.xyz.com

Router(config-gk-lcftrigger)#

remote-ext-address e164 1408....

Router(config-gk-lcftrigger#

exit

If the LCF registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger lcf alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_lcftrigger)#

remote-ext-address e164 1800....

Router(config-gk_lcftrigger)#

exit

then gatekeeper "alpha" checks all incoming LCF messages for the destination H.323 ID, e-mail ID, or remote extension address 1408 before checking for the remote extension address 1800 (for example, 18005551212).

If any one of those conditions is met, the gatekeeper sends the LCF message to the GKTMP server

"Server-west".

If the second gatekeeper "alpha" LCF trigger registration had been defined with a priority 1 instead of priority

2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those LCF messages that contain a remote extension address E.164 address starting with 1800. All other LCF messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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sccp through service-type call-check server trigger lrj

server trigger lrj

To configure the location reject (LRJ) trigger statically on the gatekeeper, use the server trigger lrjcommand in gatekeeper configuration mode. Submode commands are available after entering the server trigger

lrjcommand. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger lrj gkid priority server-id server-ip-address server-port

no server trigger lrj gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

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destination -info e164 | email-id | h323-id value Use to send LRJ RAS messages containing a specified destination to the GKTMP server application.

Configure one of the following conditions:

e164 -- Destination is an E.164 address.

email -id-- Destination is an e-mail ID.

h323 -id-- Destination is an H.323 ID.

value-- Value against which to compare the destination address in the RAS messages. For

E.164 addresses, the following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

info -only

shutdown

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco

MC3810.

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Usage Guidelines

Use this command and its optional submode commands to configure the location reject (LRJ) static server trigger. The gatekeeper checks incoming gateway LRJ messages for the configured trigger information. If an incoming LRJ message contains the specified trigger information, the gatekeeper sends the LRJ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the LRJ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the LRJ messages, the gatekeeper sends all LRJ messages to the

GKTMP server application.

If the gatekeeper receives an LRJ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming LRJ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two LRJ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

LRJ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming LRJ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one LRJ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all LRJ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger lrj sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_lrjtrigger)#

exit

The following example configures an LRJ trigger registration on gatekeeper "alpha", which send to GKTMP server "Server-west" any LRJ message containing an H.323 ID "3660-gw1" or e-mail ID joe.xyz.com. All other LRJ messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger lrj alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-lrjtrigger)#

destination-info h323-id 3660-gw1

Router(config-gk-lrjtrigger)#

destination-info email-id joe.xyz.com

Router(config-gk-lrjtrigger#

exit

If the LRJ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger lrj alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_lrjtrigger)#

destination-info e164 1800....

Router(config-gk_lrjtrigger)#

exit

then gatekeeper "alpha" checks all incoming LRJ messages for the destination H.323 ID or email ID before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the LRJ message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" LRJ trigger registration had been defined with a priority 1 instead of priority

2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those LRJ messages that contain a destination E.164

address starting with 1800. All other LRJ messages would not be sent to the GKTMP server.

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Related Commands

Command server registration-port show gatekeeper servers server trigger lrj

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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sccp through service-type call-check server trigger lrq

server trigger lrq

To configure the location request (LRQ) trigger statically on the gatekeeper, use the server trigger lrqcommand in gatekeeper configuration mode. Submode commands are available after entering the server trigger

lrqcommand. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger lrq gkid priority server-id server-ip-address server-port

no server trigger lrq gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

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destination -info e164 | email-id | h323-id value Use to send LRQ RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions:

e164 -- Destination is an E.164 address.

email -id-- Destination is an e-mail ID.

h323 -id-- Destination is an H.323 ID.

value-- Value against which to compare the destination address in the RAS messages. For

E.164 addresses, the following wildcards can be used:

• A trailing series of periods, each of which represents a single character.

• A trailing asterisk, which represents one or more characters.

info -only

redirect -reason reason-number

shutdown

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to send LRQ RAS messages containing a specific redirect reason to the GKTMP server application.

reason -number--Range is from 0 to 65535.

Currently used values are the following:

• 0 -- Unknown reason.

• 1 -- Call forwarding busy or called DTE busy.

• 2 -- Call forwarded; no reply.

• 4 -- Call deflection.

• 9 -- Called DTE out of order.

• 10 -- Call forwarding by the called DTE.

• 15 -- Call forwarding unconditionally.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

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Command Default

No trigger servers are set.

Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco

MC3810.

Usage Guidelines

Use this command and its optional submode commands to configure the location request (LRQ) static server trigger. The gatekeeper checks incoming gateway LRQ messages for the configured trigger information. If an incoming LRQ message contains the specified trigger information, the gatekeeper sends the LRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the LRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the LRQ messages, the gatekeeper sends all LRQ messages to the GKTMP server application.

If the gatekeeper receives an LRQ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming LRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two LRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

LRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming LRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one LRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all LRQ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger lrq sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_lrqtrigger)#

exit

The following example configures an LRQ trigger registration on gatekeeper "alpha", which sends to GKTMP server "Server-west" any LRQ message containing an H.323 ID "3660-gw1", e-mail ID joe.xyz.com, or a redirect reason 1. Other LRQ messages are not sent to the GKTMP server application.

Router(config-gk)#

server trigger lrq alpha 1 Server-west 10.10.10.10 1751

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Router(config-gk-lrqtrigger)#

destination-info h323-id 3660-gw1

Router(config-gk-lrqtrigger)#

destination-info email-id joe.xyz.com

Router(config-gk-lrqtrigger)#

redirect-reason 1

Router(config-gk-lrqtrigger#

exit

If the LRQ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger lrq alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_lrqtrigger)#

destination-info e164 1800....

Router(config-gk_lrqtrigger)#

exit

then gatekeeper "alpha" checks all incoming LRQ messages for the destination H.323 ID, email ID, or redirect reason before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the LRQ message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" LRQ trigger registration had been defined with a priority 1 instead of priority

2, then the second server trigger definition would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those LRQ messages that contain a destination

E.164 address starting with 1800. All other LRQ messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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server trigger rai

To configure the resources available indicator (RAI) trigger statically on the gatekeeper, use the server trigger

raicommand in gatekeeper configuration mode. Submode commands are available after entering the server

trigger raicommand. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger rai gkid priority server-id server-ip-address server-port

no server trigger rai gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

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endpoint -type value

info -only

shutdown

supported -prefix value

server trigger rai

Use to send RAI RAS messages that contain a particular endpoint type to the GKTMP server application.

value --Value against which to compare the endpoint type in the RAS messages. Valid endpoint types are the following:

gatekeeper--Endpoint is an H.323

gatekeeper.

h320-gateway--Endpoint is an H.320

gateway.

mcu--Endpoint is a multipoint control unit

(MCU).

other-gateway--Endpoint is another type of gateway not specified on this list.

proxy--Endpoint is an H.323 proxy.

terminal--Endpoint is an H.323 terminal.

voice-gateway--Endpoint is a voice gateway.

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

Use to send RAI RAS messages that contain a specific supported prefix to the GKTMP server application.

value --Value against which to compare the supported prefix in the RAS messages. The possible values are any E.164 pattern used as a gateway technology prefix. The value string can contain any of the following: 0123456789#*.

Command Default

No trigger servers are set.

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Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series,

Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.

The irrtrigger was added.

Usage Guidelines

Use this command and its optional submode commands to configure the resources available indicator (RAI) static server trigger. The gatekeeper checks incoming gateway RAI messages for the configured trigger information. If an incoming RAI message contains the specified trigger information, the gatekeeper sends the

RAI message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the RAI message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the RAI messages, the gatekeeper sends all RAI messages to the

GKTMP server application.

If the gatekeeper receives an RAI trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming RAI RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two RAI trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

RAI trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming RAI messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one RAI trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all RAI messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger rai sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_raitrigger)#

exit

The following example configures an RAI trigger registration on gatekeeper "alpha", which sends to the

GKTMP server "Server-west" any RAI message that contain an MCU endpoint, an H.323 proxy endpoint, or a supported prefix 1#. All other RAI messages are not sent to the GKTMP server.

Router(config-gk)#

server trigger rai alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-raitrigger)#

endpoint-type mcu

Router(config-gk-raitrigger)#

endpoint-type proxy

Router(config-gk-raitrigger)#

supported-prefix 1#

Router(config-gk-raitrigger#

exit

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If the RAI registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger rai alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_raitrigger)#

supported-prefix 1234*

Router(config-gk_raitrigger)#

exit

Then gatekeeper "alpha" checks all incoming RAI messages for the MCU or H.323 proxy endpoint or the supported prefix 1# before checking for the supported prefix 1234*. If any one of those conditions is met, the gatekeeper sends the RAI message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" RAI trigger registration had been defined with a priority 1 instead of priority

2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those RAI messages that contain a supported prefix of 1234*. All other RAI messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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sccp through service-type call-check server trigger rrq

server trigger rrq

To configure the registration request (RRQ) trigger statically on the gatekeeper, use the server trigger

rrqcommand in gatekeeper configuration mode. Submode commands are available after entering the server

trigger rrqcommand. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger rrq gkid priority server-id server-ip-address server-port

no server trigger rrq gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority server-id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

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endpoint -type value

info -only

shutdown

supported -prefix value

server trigger rrq

Use to send RRQ RAS messages containing a particular endpoint type to the GKTMP server application.

value --Value against which to compare the endpoint-type in the RAS messages. Valid endpoint types are the following:

gatekeeper--Endpoint is an H.323

gatekeeper.

h320-gateway--Endpoint is an H.320

gateway.

mcu--Endpoint is a multipoint control unit

(MCU).

other-gateway--Endpoint is another type of gateway not specified on this list.

proxy--Endpoint is an H.323 proxy.

terminal--Endpoint is an H.323 terminal.

voice-gateway--Endpoint is a voice gateway.

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

Use to send RRQ RAS messages containing a specific supported prefix to the GKTMP server application.

value --Value against which to compare the supported prefix in the RAS messages. The possible values are any E.164 pattern used as a gateway technology prefix. The value string can contain any of the following: 0123456789#*.

Command Default

No trigger servers are set.

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Command Modes

Gatekeeper configuration (config-gk)

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco

MC3810.

Usage Guidelines

Use this command and its optional submode commands to configure the registration request (RRQ) static server trigger. The gatekeeper checks incoming gateway RRQ messages for the configured trigger information.

If an incoming RRQ message contains the specified trigger information, the gatekeeper sends the RRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the RRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the RRQ messages, the gatekeeper sends all RRQ messages to the GKTMP server application.

If the gatekeeper receives an RRQ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming RRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two RRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

RRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming RRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one RRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all RRQ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger rrq sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_rrqtrigger)#

exit

The following example configures an RRQ trigger registration on gatekeeper "alpha", which sends to the

GKTMP server "Server-west" any RRQ message containing an MCU endpoint, an H.323 proxy endpoint, or a supported prefix 1#. Other RRQ messages are not sent to the GKTMP server.

Router(config-gk)#

server trigger rrq alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-rrqtrigger)#

endpoint-type mcu

Router(config-gk-rrqtrigger)#

endpoint-type proxy

Router(config-gk-rrqtrigger)#

supported-prefix 1#

Router(config-gk-rrqtrigger#

exit

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If the RRQ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger rrq alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_rrqtrigger)#

supported-prefix 1234*

Router(config-gk_rrqtrigger)#

exit

then gatekeeper "alpha" checks all incoming RRQ messages for the MCU or H.323 proxy endpoint or the supported prefix 1# before checking for the supported prefix 1234*. If any one of those conditions is met, the gatekeeper sends the RRQ message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" RRQ trigger registration had been defined with a priority 1 instead of priority

2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those RRQ messages that contain a supported prefix of 1234*. All other RRQ messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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server trigger urq

To configure the unregistration request (URQ) trigger statically on the gatekeeper, use the server trigger

urqcommand in gatekeeper configuration mode. Submode commands are available after entering the server

trigger urqcommand. To delete a single static trigger on the gatekeeper, use the no form of this command.

To delete all static triggers on the gatekeeper, use the allform of the command.

server trigger urq gkid priority server-id server-ip-address server-port

Submode Commands: info-only shutdown

endpoint-type value

supported-prefix value

no server trigger urq gkid priority server-id server-ip-address server-port

no server trigger all all

gkid priority

server -id

server -ip-address

server -port

Deletes all CLI-configured triggers.

Local gatekeeper identifier.

Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.

ID number of the external application.

IP address of the server.

Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.

Syntax Description

After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line.

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endpoint -type value

info -only

shutdown

supported -prefix value

server trigger urq

Use to send URQ RAS messages containing a particular endpoint type to the GKTMP server application.

value --Value against which to compare the endpoint-type in the RAS messages. Valid endpoint types are the following:

gatekeeper--Endpoint is an H.323

gatekeeper.

h320-gateway--Endpoint is an H.320

gateway.

mcu--Endpoint is a multipoint control unit

(MCU).

other-gateway--Endpoint is another type of gateway not specified on this list.

proxy--Endpoint is an H.323 proxy.

terminal--Endpoint is an H.323 terminal.

voice-gateway--Endpoint is a voice gateway.

Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol

(GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.

Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.

Use to send URQ RAS messages containing a specific supported prefix to the GKTMP server application.

value --Value against which to compare the supported prefix in the RAS messages. The possible values are any E.164 pattern used as a gateway technology prefix. The value string can contain any of the following: 0123456789#*.

Command Default

No trigger servers are set.

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Command Modes

Gatekeeper configuration

Command History

Release

12.1(1)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco

MC3810.

Usage Guidelines

Use this command and its optional submode commands to configure the unregistration request (URQ) static server trigger. The gatekeeper checks incoming gateway URQ messages for the configured trigger information.

If an incoming URQ message contains the specified trigger information, the gatekeeper sends the URQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the URQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application.

If no submode commands are configured for the URQ messages, the gatekeeper sends all URQ messages to the GKTMP server application.

If the gatekeeper receives a URQ trigger registration message that contains several trigger conditions, the conditions are treated as "OR" conditions. In other words, if an incoming URQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server.

If the gatekeeper receives two URQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two

URQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming URQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one URQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.

The the no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.

Examples

The following example configures a trigger registration on gatekeeper "sj.xyz.com" to send all URQ messages to GKTMP server "Server-123":

Router(config-gk)#

server trigger urq sj.xyz.com 1 Server-123 1.14.93.130 1751

Router(config-gk_urqtrigger)#

exit

The following example configures a URQ trigger registration on gatekeeper "alpha", which sends to the

GKTMP server "Server-west" any URQ message containing an MCU endpoint, an H.323 proxy endpoint, or a supported prefix 1#. Other URQ messages are not sent to the GKTMP server.

Router(config-gk)#

server trigger urq alpha 1 Server-west 10.10.10.10 1751

Router(config-gk-urqtrigger)#

endpoint-type mcu

Router(config-gk-urqtrigger)#

endpoint-type proxy

Router(config-gk-urqtrigger)#

supported-prefix 1#

Router(config-gk-urqtrigger#

exit

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If the URQ registration message defined above for gatekeeper "alpha" is configured and the gatekeeper receives the following trigger registration:

Router(config-gk)#

server trigger urq alpha 2 Server-west 10.10.10.10 1751

Router(config-gk_urqtrigger)#

supported-prefix 1234*

Router(config-gk_urqtrigger)#

exit

then gatekeeper "alpha" checks all incoming URQ messages for the MCU or H.323 proxy endpoint or the supported prefix 1# before checking for the supported prefix 1234*. If any one of those conditions is met, the gatekeeper sends the URQ message to the GKTMP server "Server-west".

If the second gatekeeper "alpha" URQ trigger registration had been defined with a priority 1 instead of priority

2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper

"alpha" would send to GKTMP server "Server-west" only those URQ messages that contain a supported prefix of 1234*. All other URQ messages would not be sent to the GKTMP server.

Related Commands

Command server registration-port show gatekeeper servers

Description

Configures the server listening port on the gatekeeper.

Displays the triggers configured on the gatekeeper.

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service

To load and configure a specific, standalone application on a dial peer, use the service command in application configuration mode. To remove the application from the dial peer, use the no form of this command.

service [alternate| default] service-name location

no service [alternate| default] service-name location

Syntax Description alternate default

service name location

(Optional) Alternate service to use if the service that is configured on the dial peer fails.

(Optional) Specifies that the default service

("DEFAULT") on the dial peer is used if the alternate service fails.

Name that identifies the voice application. This is a user-defined name and does not have to match the script name.

Directory and filename of the Tcl script or VoiceXML document in URL format. For example, the following are valid locations:

• built-in applications (builtin:filename)

• flash memory (flash:filename)

• HTTP server (http://../filename)

• HTTPS (HTTP over Secure Socket Layer

(SSL)) server (https://../filename)

• TFTP server (tftp://../filename)

Command Default

The default service ("DEFAULT") is used if no other services are configured.

Command Modes

Application configuration (config-app)

Command History

Release

12.3(14)T

12.4(15)T

Modification

This command was introduced.

The location argument was modified to accept an HTTPS server URL. The description of the location argument was modified to describe how to specify the location for built-in applications.

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Usage Guidelines

Use this command to load a service on the gateway. A service is a standalone application, such as a VoiceXML document or a Tcl script.

Examples

The following example shows a debitcard application configured on the dial peer.

Router(config)#

application

Router(config-app)#

service debitcard tftp://server-1//tftpboot/scripts/app_debitcard.2.0.2.8.tcl

The following example shows the VoiceXML application myapp located on an HTTPS server configured on the dial peer.

Router(config)#

application

Router(config-app)#

service myapp https://myserver/myfile.vxml

The following example shows the auto-attendant (AA) service, called aa, which is a Tcl script embedded in the Cisco IOS software.

Router(config)#

application

Router(config-app)#

service queue builtin:app-b-acd

Related Commands

Command application (application configuration) call application alternate call application voice

Description

Configures an application on a dial peer.

Specifies an alternate application to use if the application that is configured in the dial peer fails.

Defines the name of a voice application and specifies the location of the Tcl or VoiceXML document to load for this application.

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service dsapp

To configure supplementary IP Centrex-like services for FXS phones on voice gateways to interwork with

SIP-based softswitches, use the service dsappcommand in the gateway-application configuration mode.

Hookflash triggers a supplementary feature based on the current state of the call. To reset to the defaults, use the no form of this command.

service dsapp [paramspace dialpeer dial-peer tag] [paramspace disc-toggle-time seconds] [paramspace

callWaiting TRUE| FALSE] [paramspace callConference TRUE| FALSE] [paramspace blind-xfer-wait-time

seconds] [paramspace callTransfer TRUE| FALSE]

no service dsapp

Syntax Description

paramspace

dialpeer dial-peer tag

disc-toggle-time seconds

callWaiting TRUE | FALSE

callConference TRUE | FALSE

blind-xfer-wait-time seconds

callTransfer TRUE | FALSE

Defines a package or service on the gateway, the parameters in that package or service become available for configuration when you use this argument.

(Optional) Specifies the fixed dialpeer used to setup the call to the SIP server (trunk) side.

(Optional) Specifies the seconds to wait before switching to a call on hold if the active call disconnects. You can specify a range between 10 and

30 seconds.

Toggles support for call waiting.

Toggles support for call conferencing used to establish two calls with a single connection such that all three parties can talk together.

Specifies the seconds to wait before triggering a blind call transfer. You can specify a range between 0 and

10 seconds. If you specify 0 seconds, no blind transfer call occurs.

Toggles support for call transfers.

Command Default

If no supplementary features are defined, the defaults are as follows:

dialpeer : -1

disc-toggle-time : 10 seconds

callWaiting : TRUE (enabled)

callConference : TRUE (enabled)

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blind-xfer-wait-time : 0 seconds

callTransfer : TRUE (enabled)

Command Modes

Gateway-application configuration (config-app-param)

Command History

Release

12.4(11)T

Modification

This command was introduced.

Usage Guidelines

Use the service dsappcommand to configure supplementary Centrex-like features on FXS phones to interwork with SIP-based softswitches. Hookflash triggers supplementary features based on the current state of the call:

• Call Hold

• Call Waiting

• Call Transfer

• 3-Way Conference

Call Hold

Allows a call to be placed in a non-active state (with no media exchange). The table below summarizes the hookflash feature support for Call Hold.

Table 1: Call Hold Hookflash Services

State

Active call

Call on hold

Call on hold and active call

Action

Hookflash

Hookflash

Hookflash

On hook

Result

Held call for remote party.

Response to FXS Line

Second dial tone for FXS phone.

Active call.

FXS line connects to call.

Active and held calls are swapped.

FXS line connects to previous held call.

Active call is dropped.

Reminder ring on FXS line.

Call on hold goes on hook Call on hold is dropped.

None.

Active call goes on hook Active call is dropped Silence.

Call Waiting

Allows a second call to be received while the phone is active with a call. The table below summarizes the hookflash feature support for Call Waiting.

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Table 2: Call Waiting Hookflash Services

State

Active call and waiting call

Action

Hookflash.

Result

Swap active call and waiting call.

Active call goes on hook.

Active call is disconnected.

Waiting call goes on hook.

On hook.

Stay connected to active call.

Active call is dropped.

Response to FXS Line

FXS line connects to waiting call.

Silence.

None.

Reminder ring on FXS line.

Call Transfer

With call transfer, you can do the following:

• Put an active call on hold while establishing a second call.

• Set up a call between two users

• Transfer calls by using these options

• -Blind transfer

• Semi-attended transfer

• Attended transfer

The table below summarizes the hookflash feature support for Call Transfer.

Table 3: Call Transfer Hookflash Services

State

Active call

Action

Hookflash.

Call on hold and outgoing dialed or alerting or active call

On hook.

Result

Call is placed on hold.

Call on hold and active call.

Response to FXS Line

Second dial tone.

Call on hold and outgoing active call

Active call goes on hook.

Held call remains; active call dropped.

Silence.

Call on hold and outgoing active call

Call on hold goes on hook.

Active call remains; call on hold dropped.

None.

Call on hold and outgoing alerting call

Hookflash.

Active call dropped.

FXS line connects to previous held call.

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3-Way Conference

Establishes two calls with a single connection, so that three parties can talk together. The table below summarizes the hookflash feature support for 3-way conferencing.

Table 4: 3-Way Conference Hookflash Services

State

Active call

Call on hold and active call

Action

Hookflash

Result

Call on hold.

Join call on hold and active call.

Response to FXS Line

Second dial tone.

Media mixing of both calls.

Examples

Examples

Examples

You can configure DSApp services either to a specific dial-peer, or globally to all dial peers. The following example shows the configuration to enable DSApp on a specific dial peer:

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(conf-app)#

dial-peer voice 1000 pots

Gateway(config-app)#

service dsapp

The following example shows the configuration to enable DSApp globally on all dial peers:

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

global

Gateway(config-app-global)#

service default dsapp

The following example shows the configuration to enable the Call Hold feature:

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

service dsapp

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Examples

Examples

Examples

Examples

Gateway

(config-app-param)#

param callHold TRUE

The following example shows the configuration to enable the Call Waiting feature:

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

service dsapp

Gateway

(config-app-param)#

param callWaiting TRUE

The following example shows the configuration to enable the Call Transfer feature:

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

service dsapp

Gateway

(config-app-param)#

param callTransfer TRUE

The following example shows the configuration to enable the 3-Way Conferencing feature:

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

service dsapp

Gateway

(config-app-param)#

param callConference TRUE

In this example, a disconnect toggle time is configured that specifies the amount of time in seconds the system should wait before committing the call transfer after the originating call is placed on hook.

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

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Examples

Examples service dsapp

Gateway(config-app-param)#

param disc-toggle-time 10

In this example, a blind transfer call wait time is configured that specifies the amount of time in seconds the system should wait before committing the call transfer, after the originating call is placed on hook.

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

service dsapp

Gateway(config-app-param)#

param blind-xfer-wait-time 10

In this example, a fixed dial peer is configured to set up a call to the SIP server (trunk) side.

Gateway#

configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)#

application

Gateway(config-app)#

service dsapp

Gateway(config-app-param)#

param dialpeer 5000

Related Commands

Command offer call-hold

Description

Specifies the method of call hold on the gateway.

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service-flow primary upstream

To assign a quality of service (QoS) policy to the data traveling between the cable modem and the multiple service operator (MSO) cable modem termination system (CMTS), use the service-flow primary upstream command in interface configuration mode. To disable the QoS policy, use the no form of this command.

service-flow primary upstream no service-flow primary upstream

Syntax Description

This command has no arguments or keywords.

Command Default

This command is disabled by default.

Command Modes

Interface configuration (config-if)

Command History

Release

12.4(11)T

Modification

This command was introduced.

Usage Guidelines

This command is supported in the upstream direction only. Service flows are unidirectional.

Examples

The following example assigns a QoS policy to the data traveling between the cable modem and the MSO

CMTS:

Router#

configure terminal

Router(config)#

interface Cable-Modem 0/2/0

Router(config-if)#

service-flow primary upstream

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service-map

To configure the HTTP application service map for the phone proxy instance, use the service-map command in phone proxy configuration mode. To remove the HTTP application service map, use the no form of the command.

service-map server-addr ipv4 http-ipv4-address port http-server-port-number acc-addr ipv4

access-ipv4-addressport access-port-number

no service-map server-addr ipv4 http-ipv4-address port http-server-port-number acc-addr ipv4

access-ipv4-addressport access-port-number

Syntax Description

http-ipv4-address

port http-server-port-number

acc-addr ipv4 access-ipv4-address

port access-port-number

Specifies the IPv4 address of the HTTP server.

The port number of the HTTP server. The range is 1 to 65535.

Specifies the IPv4 address of the access side server.

The port number of the access side server. The range is 1 to

65535.

Command Default

No HTTP application service map is configured.

Command Modes

Phone proxy configuration mode (config-phone-proxy)

Command History

Release

15.3(3)M

Modification

This command was introduced.

Usage Guidelines

Examples

The following example shows how to configure an HTTP application service map for the phone proxy instance called “first-pp”:

Device(config)# voice-phone-proxy first-pp

Device(config-phone-proxy)# service-map server-addr ipv4 192.0.2.50 port 8080 acc-addr ipv4

10.0.0.8 port 1234

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service-relationship

To enter Annex G neighbor configuration mode and enable service relationships for the particular neighbor, use the service-relationship command in Annex G neighbor configuration mode. To exit this mode, use the

no form of this command.

service-relationship no service-relationship

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled

Command Modes

Annex G neighbor configuration (config-annexg-neigh)

Command History

Release

12.2 (11)T

Modification

This command was introduced.

Usage Guidelines

Service relationships are defined to be unidirectional. If a service relationship is established between border element A and border element B, A is entitled to send requests to B and to expect responses. For B to send requests to A and to expect responses, a second service relationship must be established. Repeat this command for each border-element neighbor that you configure.

Note

The no shutdown command must be used to enable each service relationship.

Examples

The following example enables a service relationship on a border element:

Router(config-annexg-neigh)#

service-relationship

Related Commands

Command

access -policy

inbound ttl

outbound retry -interval

Description

Requires that a neighbor be explicitly configured.

Sets the inbound time-to-live value.

Defines the retry period for attempting to establish the outbound relationship between border elements.

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Command retry interval retry window shutdown service-relationship

Description

Defines the time between delivery attempts.

Defines the total time for which a border element attempts delivery.

Enables or disables the border element.

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service-type call-check

To identify preauthentication requests to the authentication, authorization, and accounting (AAA) server, use the service-type call-check command in AAA preauthentication configuration mode. To return this setting to the default, use the no form of this command.

service-type call-check no service-type call-check

Syntax Description

This command has no arguments or keywords.

Command Default

The service type is not set to call-check.

Command Modes

AAA preauthentication configuration (config-preauth)

Command History

Release

12.2(11)T

Modification

This command was introduced.

Usage Guidelines

Setting the service-type attribute to call-check causes preauthentication access requests to include this value, which allows AAA servers to distinguish preauthentication requests from other types of Access-Requests.

This command has no effect on packets that are not of the preauthentication type.

Examples

The following example sets the RADIUS service-type attribute to call-check:

Router(config)#

aaa preauth

Router(config-preauth)#

service-type call-check

Related Commands

Command aaa preauth

Description

Enters AAA preauthentication configuration mode.

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session, page 109

session group, page 111

session protocol (dial peer), page 113

session protocol (Voice over Frame Relay), page 115

session protocol aal2, page 117

session protocol multicast, page 118

session refresh, page 119

session start, page 120

session target (MMoIP dial peer), page 122

session target (POTS dial peer), page 126

session target (VoATM dial peer), page 128

session target (VoFR dial peer), page 132

session target (VoIP dial peer), page 135

session target, page 141

session transport, page 142

session transport (H.323 voice-service), page 144

session transport (SIP), page 146

session-set, page 148

session-timeout, page 150

set, page 151

set http client cache stale, page 153

set pstn-cause, page 154

set sip-status, page 157

settle-call, page 160

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settlement, page 162

settlement roam-pattern, page 164

sgcp, page 166

sgcp call-agent, page 168

sgcp graceful-shutdown, page 170

sgcp max-waiting-delay, page 172

sgcp modem passthru, page 174

sgcp quarantine-buffer disable, page 176

sgcp request retries, page 178

sgcp request timeout, page 180

sgcp restart, page 182

sgcp retransmit timer, page 184

sgcp timer, page 186

sgcp tse payload, page 188

source filter, page 190

session through sgcp tse payload

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session

To associate a transport session with a specified session group, use the session command in backhaul session manager configuration mode. To delete the session, use the no form of this command.

session group group-name remote-ip remote-port local-ip local-port priority

no session group group-name remote-ip remote-port local-ip local-port priority

Syntax Description

group -name

remote -ip

remote -port

local -ip

local -port

priority

Session-group name.

Remote IP address.

Remote port number. Range is from 1024 to 9999.

Local IP address.

Local port number. Range is from 1024 to 9999.

Priority of the session-group. Range is from 0 to 9999;

0 is the highest priority.

Command Default

No default behavior or values

Command Modes

Backhaul session manager configuration

Command History

Release

12.1(1)T

12.2(4)T

12.2(2)XB

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was introduced.

This command was implemented on the following platforms: Cisco 2600 series,

Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was implemented on the Cisco IAD2420 series. Support for the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco

AS5850 in this release.

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Usage Guidelines

It is assumed that the server is located on a remote machine.

Examples

The following example associates a transport session with the session group "group5" and specifies the parameters:

Router(config-bsm)#

session group group5

172.13.2.72 5555 172.18.72.198 5555 1

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session group

To associate a transport session with a specified session group, use the session groupcommand in backhaul session-manager configuration mode. To delete the session, use the no form of this command.

session group group-name remote-ip remote-port local-ip local-port priority

no session group group-name remote-ip remote-port local-ip local-port priority

Syntax Description

group -name

remote -ip

remote -port

local -ip

local -port

priority

Session-group name.

Remote IP address.

Remote port number. Range is from 1024 to 9999.

Local IP address.

Local port number. Range is from 1024 to 9999.

Priority of the session group. Range is from 0 to 9999;

0 has the highest priority.

Command Default

No default behavior or values.

Command Modes

Backhaul session-manager configuration

Command History

Release

12.1(1)T

12.2(2)T

12.2(4)T

12.2(2)XB1

12.2(8)T

Modification

This command was introduced.

This command was implemented on the Cisco 7200 series.

This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5850.

This command was implemented on the Cisco IAD2420 series.

Usage Guidelines

The server is assumed to be located on a remote machine.

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Examples

The following example associates a transport session with the session group named "group5" and specifies the keywords described above:

session group group5

172.16.2.72 5555 192.168.72.198 5555 1

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session protocol (dial peer)

To specify a session protocol for calls between local and remote routers using the packet network, use the

session protocol command in dial-peer configuration mode. To reset to the default, use the no form of this command.

session protocol {aal2-trunk| cisco| sipv2| smtp}

no session protocol

Syntax Description

aal2 -trunk

cisco sipv2

smtp

Dial peer uses ATM adaptation layer 2 (AAL2) nonswitched trunk session protocol.

Dial peer uses the proprietary Cisco VoIP session protocol.

Dial peer uses the Internet Engineering Task Force

(IETF) Session Initiation Protocol (SIP). Use this keyword with the SIP option.

Dial peer uses Simple Mail Transfer Protocol (SMTP) session protocol.

Command Default

No default behaviors or values

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

12.0(3)XG

12.0(4)XJ

12.1(1)XA

12.1(1)T

12.1(2)T

12.2(2)T

Modification

This command was introduced for VoIP peers on the Cisco 3600 series.

This command was modified to support VoFR) dial peers.

This command was modified for store-and-forward fax on the Cisco AS5300.

This command was implemented for VoATM dial peers on the Cisco MC3810.

The aal2-trunk keyword was added.

This command was integrated into Cisco IOS Release 12.1(1)T. The sipv2 keyword was added.

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was implemented on the Cisco 7200 series.

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Release

12.2(4)T

12.2(2)XA

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was implemented on the Cisco 1750.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco

AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release. The

aal2-trunk and smtp keywords are not supported on the Cisco 7200 series in this release.

This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco

AS5850 in this release.

Usage Guidelines

The ciscokeyword is applicable only to VoIP on the Cisco 1750, Cisco 1751, Cisco 3600 series, and Cisco

7200 series routers.

The aal2-trunk keyword is applicable only to VoATM on the Cisco 7200 series router.

This command applies to both on-ramp and off-ramp store-and-forward fax functions.

Examples

The following example shows that AAL2 trunking has been configured as the session protocol: dial-peer voice 10 voatm session protocol aal2-trunk

The following example shows that Cisco session protocol has been configured as the session protocol: dial-peer voice 20 voip session protocol cisco

The following example shows that a VoIP dial peer for SIP has been configured as the session protocol for

VoIP call signaling: dial-peer voice 102 voip session protocol sipv2

Related Commands

Command dial-peer voice session target (VoIP)

Description

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

Configures a network-specific address for a dial peer.

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session protocol (Voice over Frame Relay)

To establish a Voice over Frame Relay protocol for calls between the local and remote routers via the packet network, use the session protocol command in dial-peer configuration mode. To reset to the default, use the

no form of this command.

session protocol {cisco-switched| frf11-trunk}

no session protocol

Syntax Description

cisco -switched

frf11 -trunk

Proprietary Cisco VoFR session protocol. (This is the only valid session protocol for the Cisco 7200 series.)

FRF.11 session protocol.

Command Default cisco-switched

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

12.0(3)XG

12.0(4)T

Modification

This command was introduced for VoIP.

This command was modified to support VoFR on the following platforms:

Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.

The cisco-switched and frf11-trunk keywords were added for VoFR dial peers.

Usage Guidelines

For Cisco-to-Cisco dial peer connections, Cisco recommends that you use the default session protocol because of the advantages it offers over a pure FRF.11 implementation. When connecting to FRF.11-compliant equipment from other vendors, use the FRF.11session protocol.

Note

When using the FRF.11 session protocol, you must also use the called-number command.

Examples

The following example configures the FRF.11 session protocol for VoFR dial peer 200: dial-peer voice 200 vofr

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session through sgcp tse payload session protocol (Voice over Frame Relay)

session protocol frf11-trunk called-number 5552150

Related Commands

Command called-number (dial-peer) codec (dial-peer) cptone destination-pattern dtmf-relay (Voice over Frame Relay) preference session target signal-type

Description

Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11

trunk connection.

Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.

Specifies a regional analog voice interface-related tone, ring, and cadence setting.

Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.

Enables the generation of FRF.11 Annex A frames for a dial peer.

Indicates the preferred order of a dial peer within a rotary hunt group.

Specifies a network-specific address for a specified dial peer or destination gatekeeper.

Sets the signaling type to be used when connecting to a dial peer.

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session through sgcp tse payload session protocol aal2

session protocol aal2

To enter voice-service-session configuration mode and specify ATM adaptation layer 2 (AAL2) trunking, use the session protocol aal2command in voice-service configuration mode.

session protocol aal2

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values.

Command Modes

Voice-service configuration (config-voi-serv)

Command History

Release

12.1(1)XA

12.1(2)T

12.2(2)T

Modification

This command was introduced on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was implemented on the Cisco 7200 series.

Usage Guidelines

This command applies to VoATM on theCisco 7200 series router.

In the voice-service-session configuration mode for AAL2, you can configure only AAL2 features, such as call admission control and subcell multiplexing.

Examples

The following example accesses voice-service-session configuration mode, beginning in global configuration mode: voice service voatm session protocol aal2

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session through sgcp tse payload session protocol multicast

session protocol multicast

To set the session protocol as multicast, use the session protocol multicast command in dial-peer configuration mode. To reset to the default protocol, use the no version of this command.

session protocol multicast no session protocol multicast

Syntax Description

This command has no arguments or keywords.

Command Default

Default session protocol: Cisco.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

12.1(2)XH

12.1(3)T

12.2(8)T

Modification

This command was introduced for the Cisco Hoot and Holler over IP application on the Cisco 2600 series and Cisco 3600 series.

This command was integrated into Cisco IOS Release 12.1(3)T.

This command was implemented on the Cisco 1750 and Cisco 1751.

Usage Guidelines

Use this command for voice conferencing in a hoot and holler networking implementation. This command allows more than two ports to join the same session simultaneously.

Examples

The following example shows the use of the session protocol multicast dial-peer configuration command in context with its accompanying commands: dial-peer voice 111 voip destination-pattern 111 session protocol multicast session target ipv4:237.111.0.111:22222 ip precedence 5 codec g711ulaw

Related Commands

Command session target ipv4

Description

Assigns the session target for voice-multicasting dial peers.

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session through sgcp tse payload session refresh

session refresh

To enable SIP session refresh globally, use the session refreshcommand in SIP configuration mode. To disable the session refresh, use the no form of this command.

session refresh no session refresh

Syntax Description

This command has no arguments or keywords.

Command Default

No session refresh

Command Modes

SIP configuration (conf-serv-sip)

Command History

Release

15.1(2)T

Modification

This command was introduced.

Usage Guidelines

Use the SIP session refreshcommand to send the session refresh request.

Examples

The following example sets the session refresh under SIP configuration mode:

Router(conf-serv-sip)#

Session refresh

Related Commands

Command voice-class sip session refresh

Description

Enables session refresh at dial-peer level.

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session through sgcp tse payload session start

session start

To start a new instance (session) of a Tcl IVR 2.0 application, use the session start command in application configuration mode. To stop the session and remove the configuration, use the no form of this command.

session start instance-name application-name

no session start instance-name

Syntax Description

instance-name application-name

Alphanumeric label that uniquely identifies this application instance.

Name of the Tcl application. This is the name of the application that was assigned with the service command.

Command Default

No default behavior or values

Command Modes

Application configuration

Command History

Release

12.3(14)T

Modification

This command was introduced to replace the call application session start

(global configuration) command.

Usage Guidelines

• This command starts a new session, or instance, of a Tcl IVR 2.0 application. It cannot start a session for a VoiceXML application because Cisco IOS software cannot start a VoiceXML application without an active call leg.

• You can start an application instance only after the Tcl application is loaded onto the gateway with the

service command.

• If this command is used, the session restarts if the gateway reboots.

• If the application session stops running, it does not restart unless the gateway reboots. A Tcl script might intentionally stop running by executing a "call close" command for example, or it might fail because of a script error.

• You can start multiple instances of the same application by using different instance names.

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session through sgcp tse payload session start

Examples

The following example starts a session named my_instance for the application named demo: application session start my_instance demo

The following example starts another session for the application named demo: application session start my_instance2 demo

Related Commands

Command Description call application session start (global configuration)

Starts a new instance (session) of a Tcl IVR 2.0

application.

service

Loads a specific, standalone application on a dial peer.

show call application services registry show call application sessions

Displays a one-line summary of all registered services.

Displays summary or detailed information about voice application sessions.

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session through sgcp tse payload session target (MMoIP dial peer)

session target (MMoIP dial peer)

To designate an e-mail address to receive T.37 store-and-forward fax calls from a Multimedia Mail over IP

(MMoIP) dial peer, use the session targetcommand indial peer configuration mode. To remove the target address, use the no form of this command.

session target mailto: {name| $d$| $m$| $e$} [@domain-name]

no session target

Syntax Description mailto:

name

$d$

$m$

Matching calls are passed to the network using Simple

Mail Transfer Protocol (SMTP) or Extended Simple

Mail Transfer Protocol (ESMTP).

String that can be an e-mail address, name, or mailing list alias.

Macro that is replaced by the destination pattern of the gateway access number, which is the called number or dialed number identification service

(DNIS) number.

Macro that is replaced by the redirecting dialed number (RDNIS) if present; otherwise, it is replaced by the gateway access number (DNIS). This macro requires use of the fax detection interactive voice response (IVR) application.

Note

Other strings can be passed to mailto in place of $m$ if you modify the fax detection application Tool Command Language (Tcl) script or VoiceXML document. For more information, see to the readme file that came with the Tcl script or the Cisco VoiceXML

Programmers Guide.

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$e$

@ domain-name

Macro that is replaced by the DNIS, the RDNIS, or a string that represents a valid e-mail address, as specified by the cisco-mailtoaddress variable in the transfer tag of the VoiceXML fax detection document.

By default, if the cisco-mailtoaddress variable is not specified in the fax detection document, the DNIS is mapped to $e$.

If $e$ is not specified for the session target mailto command in the MMoIP dial peer, but the

cisco-mailtoaddres s variable is specified in the transfer tag of the fax detection document, then whatever is specified in the MMoIP dial peer takes precedence; the cisco-mailtoaddress variable is ignored.

Note

If a domain name is configured with this command, the VoiceXML document should pass only the username portion of the e-mail address and not the domain. If the domain name is passed from cisco-mailtoaddress, the session target mailto command should specify only $e$.

(Optional) String that contains the domain name to be associated with the target address, preceded by the at sign (@); for example, @mycompany.com .

Command Default

No default behavior or values

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

12.0(4)T

12.1(5)XM1

12.2(2)XA

12.2(2)XB

Modification

This command was introduced.

This command was modified to support store-and-forward fax.

The $m$keyword was introduced for the fax detection feature on the Cisco

AS5300.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

The $e$keyword was introduced for VoiceXML fax detection on the Cisco

AS5300.

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session through sgcp tse payload session target (MMoIP dial peer)

Release

12.2(8)T

12.2(11)T

Modification

This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco

3600 series, Cisco 3725, and Cisco 3745.

This command was implemented on the following platforms: Cisco AS5300,

Cisco AS5350, and Cisco AS5400.

Usage Guidelines

Use this command to deliver e-mail to one recipient by specifying one e-mail name, or to deliver e-mail to multiple recipients by specifying an e-mail alias as the name argument and having that alias expanded by the mailer.

Use the $m$ macro to include the redirecting dialed number (RDNIS) as part of the e-mail name when using the fax detection IVR application. If $m$ is specified and RDNIS is not present in the call information, the access number of the gateway (the dialed number, or DNIS) is used instead. For example, if the calling party originally dialed 6015550111 to send a fax, and the call was redirected (forwarded on busy or no answer) to

6015550122 (the gateway), the RDNIS is 6015550111, and the DNIS is 6015550122.

Use the $e$ macro to map the cisco-mailtoaddress variable in the VoiceXML fax detection document to the username portion of the e-mail address when sending a fax. If the VoiceXML document does not specify the

cisco-mailtoaddress variable in the transfer tag, the application maps the DNIS to the e-mail address username.

Examples

The following example delivers fax-mail to multiple recipients: dial-peer voice 10 mmoip session target mailto:[email protected]

Assuming that mailer.example.com is running the sendmail application, you can put the following information into its /etc/aliases file: marketing-information: [email protected], [email protected]

The following example uses the fax detection IVR application. Here, the session target (MMoIP dial

peer)commandforwards fax calls to an e-mail account that uses the Redirected Dialed Number Identification

Service (RDNIS) as part of its address. In this example, the calling party originally dialed 6015550111 to send a fax, and the call was forwarded (on busy or no answer) to 6015550122, which is the incoming number for the gateway being configured. The RDNIS is 6015550111, and the dialed number (DNIS) is 6015550122.

When faxes are forwarded from the gateway, the session target in the example is expanded to

[email protected]

dial-peer voice 4 mmoip session target mailto:[email protected]

The following examples configure a session target for a VoiceXML fax detection application. In this example, the VoiceXML document passes just the username portion of the e-mail address, for example, "johnd": dial-peer voice 4 mmoip session target mailto:[email protected]

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In this example, the VoiceXML document passes the complete e-mail address including domain name, for example, "[email protected]": dial-peer voice 5 mmoip session target mailto:$e$

Related Commands

Command destination-pattern dial-peer voice

Description

Specifies either the partial or full E.164 telephone number (depending on your dial plan) used to match the dial peer.

Enters dial-peer configuration mode and defines a dial peer.

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session through sgcp tse payload session target (POTS dial peer)

session target (POTS dial peer)

To designate loopback calls from a POTS dial peer, use the session target command in dial-peer configuration mode. To reset to the default, use the no form of this command.

session target {loopback:compressed| loopback:uncompressed}

no session target

Syntax Description loopback:compressed loopback:uncompressed

All voice data is looped back in compressed mode to the source.

All voice data is looped back in uncompressed mode to the source.

Command Default

No loopback calls are designated.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

12.0(3)T

12.2(2)XA

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series.

This command was implemented on the Cisco AS5300.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco

AS5850 is not included in this release.

This command was integrated into Cisco IOS Release 12.2(11)T and is supported on the Cisco AS5200, Cisco AS5350, Cisco AS5400, and Cisco

AS5850 in this release.

Usage Guidelines

Use this command to test the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.

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session through sgcp tse payload session target (POTS dial peer)

Examples

The following example loops back the traffic from the dial peer in compressed mode: dial-peer voice 10 pots session target loopback:compressed

Related Commands

Command

dial -peer voice

Description

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

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session through sgcp tse payload session target (VoATM dial peer)

session target (VoATM dial peer)

To specify a network-specific address for a specified VoATM dial peer, use the session target command in dial-peer configuration mode. To reset to the default, use the no form of this command.

Cisco 3600 Series Routers

session target interface pvc {name| vpi/vci| vci}

no session target

Cisco 7200 Series Routers

session target atm slot/port pvc {word| vpi/vci| vci} cid

no session target

Syntax Description

serial

atm

interface

slot / port

pvc

name word

vpi / vci

vci

Serial interface for the dial-peer address.

ATM interface. The only valid number is 0.

Interface type and interface number on the router.

Slot and port numbers for the dial-peer address.

Specific ATM permanent virtual circuit (PVC) for this dial peer.

PVC name.

(Optional) Name that identifies the PVC. The argument can identify the PVC if a word identifier was assigned when the PVC was created.

ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC. Values are as follows:

• Cisco 3600 series with Multiport T1/E1 ATM network module with inverse multiplexing over

ATM (IMA): vpirange is from 0 to 5; vci range is from 1 to 255.

• OC3 ATM network module: vpi range is from

0 to 15; vci range is from 1 to 1023.

ATM network virtual channel identifier (VCI) of this

PVC.

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session through sgcp tse payload session target (VoATM dial peer)

cid ATM network channel identifier (CID) of this PVC.

Range is from 8 to 255.

Command Default

Command is enabled with no IP address or domain name defined.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

11.3(1)MA

12.0(3)XG

12.0(4)T

12.0(7)XK

12.1(1)XA

12.1(2)T

12.2(2)T

Modification

This command was introduced.

This command was modified to support VoATM, VoHDLC, and POTS dial peers. The command was implemented on the Cisco MC3810.

This command was modified to support VoFR dial peers. The command was implemented on the Cisco 2600 series and Cisco 3600 series.

This command was integrated into Cisco IOS Release 12.0(4)T.

This command was modified to support VoATM and VoIP dial peers. The command was implemented on the Cisco 3600 series and the Cisco MC3810.

Support for VoHDLC was removed.

This command was modified to provide enhanced support for VoATM dial peers.

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was implemented on the Cisco 7200 series.

Usage Guidelines

Use the session targetcommand to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol that you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.

Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.

This command applies to on-ramp store-and-forward fax functions.

You must enter the session protocol aal2-trunk dial-peer configuration command before you can specify a

CID for a dial peer for VoATM on the Cisco 7200 series router.

Note

This command does not apply to POTS dial peers.

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session through sgcp tse payload session target (VoATM dial peer)

Examples

The following example configures a session target for VoATM. The session target is sent to ATM interface

0 for a PVC with a VCI of 20.

dial-peer voice 12 voatm destination-pattern 13102221111 session target atm0 pvc 20

The following example delivers fax-mail to multiple recipients: dial-peer voice 10 mmoip session target [email protected]

Assuming that mailer.example.com is running sendmail, you can put the following information into its

/etc/aliases file: marketing-information: [email protected], [email protected]

The following example configures a session target for VoATM. The session target is sent to ATM interface

0, and is for a PVC with a VPI/VCI of 1/100.

dial-peer voice 12 voatm destination-pattern 13102221111 session target atm1/0 pvc 1/100

Related Commands

Command called-number codec (dial-peer) cptone destination-pattern dtmf-relay preference session protocol session target session target loopback

Description

Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.

Specifies the voice coder rate of speech for a dial peer.

Specifies a regional tone, ring, and cadence setting for an analog voice port.

Specifies either the prefix or full E.164 telephone number (depending on the dial plan) to be used for a dial peer.

Enables the DSP to generate FRF.11 Annex A frames for a dial peer.

Indicates the preferred selection order of a dial peer within a hunt group.

Establishes a VoFR protocol for calls between local and remote routers via the packet network.

Configures a network-specific address for a dial peer.

Tests the voice transmission path of a call.

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session through sgcp tse payload

Command signal-type session target (VoATM dial peer)

Description

Sets the signaling type to be used when connecting to a dial peer.

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session through sgcp tse payload session target (VoFR dial peer)

session target (VoFR dial peer)

To specify a network-specific address for a specified VoFR dial peer, use the session target command in dial-peer configuration mode. To reset to the default, use the no form of this command.

Cisco 2600 Series and Cisco 3600 Series Routers

session target interface dlci [ cid ]

no session target

Cisco 7200 Series Routers session target interface dlci no session target

Syntax Description

interface dlci cid

Serial interface and interface number (slot number and port number) associated with this dial peer. For the range of valid interface numbers for the selected interface type, enter a ? character after the interface type.

Data link connection identifier for this dial peer.

Range is from 16 to 1007.

(Optional) DLCI subchannel to be used for data on

FRF.11 calls. A CID must be specified only when the session protocol is frf11-trunk. When the session protocol is cisco-switched, the CID is dynamically allocated. Range is from 4 to 255.

Note

By default, CID 4 is used for data; CID 5 is used for call-control. We recommend that you select CID values between 6 and 63 for voice traffic. If the CID is greater than 63, the FRF.11 header contains an extra byte of data.

Command Default

The default for this command is enabled with no IP address or domain name defined.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

Modification

This command was introduced.

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Cisco IOS Voice Command Reference - S commands

session through sgcp tse payload session target (VoFR dial peer)

Release

11.3(1)MA

12.0(3)XG

12.0(4)T

Modification

This command was implemented for VoFR, VoHDLC, and POTS dial peers on the Cisco MC3810.

This command was implemented for VoFR dial peers on the Cisco 2600 series and Cisco 3600 series. The cid option was added.

This command was integrated into Cisco IOS Release12.0(4)T and implemented for VoFR and POTS dial peers on the Cisco 7200 series.

Usage Guidelines

Use the session targetcommand to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.

The session target loopback command is used for testing the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.

For VoFR dial peers, the cid option is not allowed when the cisco-switchedoption for the session protocol command is used.

Examples

The following example configures serial interface 1/0, DLCI 100 as the session target for Voice over Frame

Relay dial peer 200 (an FRF.11 dial peer) using the FRF.11 session protocol: dial-peer voice 200 vofr destination-pattern 13102221111 called-number 5552150 session protocol frf11-trunk session target serial 1/0 100 20

The following example delivers fax-mail to multiple recipients: dial-peer voice 10 mmoip session target [email protected]

Assuming that mailer.example.com is running sendmail, you can put the following information into its

/etc/aliases file: marketing-information: [email protected], [email protected]

Related Commands

Command called-number codec (dial-peer) cptone

Description

Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.

Specifies the voice coder rate of speech for a dial peer.

Specifies a regional tone, ring, and cadence setting for an analog voice port.

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session target (VoFR dial peer)

Command destination-pattern dtmf-relay preference session protocol signal-type session through sgcp tse payload

Description

Specifies either the prefix or the full E.164 telephone number (depending on the dial plan) to be used for a dial peer.

Enables the DSP to generate FRF.11 Annex A frames for a dial peer.

Indicates the preferred selection order of a dial peer within a hunt group.

Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.

Sets the signaling type to be used when connecting to a dial peer.

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session through sgcp tse payload session target (VoIP dial peer)

session target (VoIP dial peer)

To designate a network-specific address to receive calls from a VoIP or VoIPv6 dial peer, use the session

target command in dial peer configuration mode. To reset to the default, use the no form of this command.

Cisco 1751, Cisco 3725, Cisco 3745, and Cisco AS5300

session target {dhcp| ipv4:destination-address| ipv6:[destination-address]| dns:[$s$.| $d$.| $e$.| $u$.]

hostname| enum:table-num| loopback:rtp| ras| sip-server| registrar} [:port]

no session target

Cisco 2600 Series, Cisco 3600 Series, Cisco AS5350, Cisco AS5400, and Cisco AS5850

session target {dhcp| ipv4:destination-address| ipv6:[destination-address]| dns:[$s$.| $d$.| $e$.| $u$.]

hostname| enum:table-num| loopback:rtp| ras| settlement provider-number| sip-server| registrar} [:port]

no session target

Syntax Description dhcp

ipv4: destination -address

ipv6: [destination-address]

Configures the router to obtain the session target via

DHCP.

Note

The dhcp option can be made available only if the Session Initiation Protocol (SIP) is used as the session protocol. To enable SIP, use the session protocol (dial peer) command.

Configures the IP address of the dial peer to receive calls. The colon is required.

Configures the IPv6 address of the dial peer to receive calls. Square brackets must be entered around the

IPv6 address. The colon is required.

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session target (VoIP dial peer) dns:[$s$]

hostname

enum: table -num

loopback:rtp ras

sip -server

: port

settlement provider -number

registrar

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Configures the host device housing the domain name system (DNS) server that resolves the name of the dial peer to receive calls. The colon is required.

Use one of the following macros with this keyword when defining the session target for VoIP peers:

$s$. --(Optional) Source destination pattern is used as part of the domain name.

$d$. --(Optional) Destination number is used as part of the domain name.

$e$. --(Optional) Digits in the called number are reversed and periods are added between the digits of the called number. The resulting string is used as part of the domain name.

$u$. --(Optional) Unmatched portion of the destination pattern (such as a defined extension number) is used as part of the domain name.

hostname --String that contains the complete hostname to be associated with the target address; for example, serverA.example1.com.

Configures ENUM search table number. Range is from 1 to 15. The colon is required.

Configures all voice data to loop back to the source.

The colon is required.

Configures the registration, admission, and status

(RAS) signaling function protocol. A gatekeeper is consulted to translate the E.164 address into an IP address.

Configures the global SIP server is the destination for calls from the dial peer.

(Optional) Port number for the dial-peer address. The colon is required.

Configures the settlement server as the target to resolve the terminating gateway address.

• The provider-number argument specifies the provider IP address.

Specifies to route the call to the registrar end point.

• The registrar keyword is available only for SIP dial peers.

session through sgcp tse payload session target (VoIP dial peer)

Command Default

No IP address or domain name is defined.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release

11.3(1)T

12.0(3)T

12.0(4)XJ

12.1(1)T

12.2(2)XA

12.2(2)XB1

12.2(8)T

12.2(11)T

12.4(22)T

12.4(22)YB

15.0(1)M

15.1(3)T

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series.

This command was modified. This command was implemented on the Cisco

AS5300. The ras keyword was added.

This command was implemented for store-and-forward fax on the Cisco

AS5300.

This command was integrated into Cisco IOS Release 12.1(1)T. The settlement and sip-server keywords were added.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was not included in this release.

This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco

AS5400, and Cisco AS5850. The enum keyword was added.

This command was modified. Support for IPv6 was added.

This command was modified. The dhcp keyword was added.

This command was integrated into Cisco IOS Release 15.0(1)M.

This command was modified. The registrar keyword was added.

Usage Guidelines

Use the session target command to specify a network-specific destination for a dial peer to receive calls from the current dial peer. You can select an option to define a network-specific address or domain name as a target, or you can select one of several methods to automatically determine the destination for calls from the current dial peer.

Use the session target dns command with or without the specified macros. Using the optional macros can reduce the number of VoIP dial-peer session targets that you must configure if you have groups of numbers associated with a particular router.

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session through sgcp tse payload session target (VoIP dial peer)

Examples

The session target enum command instructs the dial peer to use a table of translation rules to convert the dialed number identification service (DNIS) number into a number in E.164 format. This translated number is sent to a DNS server that contains a collection of URLs. These URLs identify each user as a destination for a call and may represent various access services, such as SIP, H.323, telephone, fax, e-mail, instant messaging, and personal web pages. Before assigning the session target to the dial peer, configure an ENUM match table with the translation rules using the voice enum-match-tablecommand in global configuration mode. The table is identified in the session target enum command with the table-num argument.

Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin.

Use the session target dhcp command to specify that the session target host is obtained via DHCP. The dhcp option can be made available only if the SIP is being used as the session protocol. To enable SIP, use the

session protocol(dial peer) command.

In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command.

For the session target settlement provider-number command, when the VoIP dial peers are configured for a settlement server, the provider-number argument in the session target and settle-call commands should be identical.

Use the session target sip-server command to name the global SIP server interface as the destination for calls from the dial peer. You must first define the SIP server interface by using the sip-server command in

SIP user-agent (UA) configuration mode. Then you can enter the session target sip-server option for each dial peer instead of having to enter the entire IP address for the SIP server interface under each dial peer.

After the SIP endpoints are registered with the SIP registrar in the hosted unified communications (UC), you can use the session target registrar command to route the call automatically to the registrar end point. You must configure the session target command on a dial pointing towards the end point.

The following example shows how to create a session target using DNS for a host named "voicerouter" in the domain example.com: dial-peer voice 10 voip session target dns:voicerouter.example.com

The following example shows how to create a session target using DNS with the optional $u$. macro. In this example, the destination pattern ends with four periods (.) to allow for any four-digit extension that has the leading number 1310555. The optional $u$. macro directs the gateway to use the unmatched portion of the dialed number--in this case, the four-digit extension--to identify a dial peer. The domain is "example.com." dial-peer voice 10 voip destination-pattern 1310555....

session target dns:$u$.example.com

The following example shows how to create a session target using DNS, with the optional $d$. macro. In this example, the destination pattern has been configured to 13105551111. The optional macro $d$. directs the gateway to use the destination pattern to identify a dial peer in the "example.com" domain.

dial-peer voice 10 voip destination-pattern 13105551111 session target dns:$d$.example.com

The following example shows how to create a session target using DNS, with the optional $e$. macro. In this example, the destination pattern has been configured to 12345. The optional macro $e$. directs the gateway

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to do the following: reverse the digits in the destination pattern, add periods between the digits, and use this reverse-exploded destination pattern to identify the dial peer in the "example.com" domain.

dial-peer voice 10 voip destination-pattern 12345 session target dns:$e$.example.com

The following example shows how to create a session target using an ENUM match table. It indicates that calls made using dial peer 101 should use the preferential order of rules in enum match table 3: dial-peer voice 101 voip session target enum:3

The following example shows how to create a session target using DHCP: dial-peer voice 1 voip session protocol sipv2 voice-class sip outbound-proxy dhcp session target dhcp

The following example shows how to create a session target using RAS: dial-peer voice 11 voip destination-pattern 13105551111 session target ras

The following example shows how to create a session target using settlement: dial-peer voice 24 voip session target settlement:0

The following example shows how to create a session target using IPv6 for a host at

2001:10:10:10:10:10:10:230a:5090: dial-peer voice 4 voip destination-pattern 5000110011 session protocol sipv2 session target ipv6:[2001:0DB8:10:10:10:10:10:230a]:5090 codec g711ulaw

The following example shows how to configure Cisco Unified Border Element (UBE) to route a call to the registering end point: dial-peer voice 4 voip session target registrar

Related Commands

Command destination-pattern

dial -peer voice session protocol (dial peer)

settle -call

Description

Specifies either the prefix or the full E.164 telephone number (depending on the dial plan) to be used for a dial peer.

Enters dial peer configuration mode and specifies the method of voice-related encapsulation.

Specifies a session protocol for calls between local and remote routers using the packet network dial peer configuration mode.

Specifies that settlement is to be used for the specified dial peer, regardless of the session target type.

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session target (VoIP dial peer)

Command

sip -server

voice enum -match-table

session through sgcp tse payload

Description

Defines a network address for the SIP server interface.

Initiates the ENUM match table definition.

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session target

To configure the Session Initiation Protocol (SIP) Uniform Resource Locator (URI) as the session target for a dial peer, use the session target sip-uri command in dial peer voice configuration mode. To disable this configuration, use the no form of the command.

session target no session target

Command Default

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Release

15.4(1)T

Modification

This command was introduced.

Usage Guidelines

Examples

The following example shows how to configure the SIP URI as the session target for a dial peer using the

session target sip-uri command:

Device>

enable

Device#

configure terminal

Device(config)#

dial-peer voice 25 voip

Device(config-dial-peer)#

session protocol sipv2

Device(config-dial-peer)#

destination uri mydesturi

Device(config-dial-peer)#

session target sip-uri

Device(config-dial-peer)#

end

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session through sgcp tse payload session transport

session transport

To configure a VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in dial-peer configuration mode. To reset to the system default keyword, use the no form of this command.

session transport {system| tcp [tls]| udp}

no session transport {system| tcp [tls]| udp}

Syntax Description

system

tcp

tls udp

The SIP dial peer defers to the voice service VoIP session transport.

The SIP dial peer uses the TCP transport layer protocol.

(Optional) The SIP dial peer uses Transport Layer

Security (TLS) over the TCP transport layer protocol.

The SIP dial peer uses the UDP transport layer protocol. This is the default.

Command Default

UDP

Note

The transport protocol specified with the transportcommand must matchthe one specified with this command.

Command Modes

Dial-peer configuration

Command History

Release

12.1(1)T

12.2(2)XA

12.2(2)XB1

12.2(11)T

12.4(6)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

The optional tls keyword was added to the command.

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Usage Guidelines

Use the show sip-ua status command to ensure that the transport protocol that you set using this command matches the protocol set using the transport command. The transport command is used in dial-peer configuration mode to specify the SIP transport method, either UDP, TCP, or TLS over TCP.

Examples

The following example shows a VoIP dial peer configured to use TCP as the underlying transport layer protocol for SIP messages: dial-peer voice 102 voip session transport tcp

The following example shows a VoIP dial peer configured to use TLS over TCP as the underlying transport layer protocol for SIP messages: dial-peer voice 102 voip session transport tcp tls

The following example shows a VoIP dial peer configured to use UDP as the underlying transport layer protocol for SIP messages: dial-peer voice 102 voip session transport udp

Related Commands

Command show sip-ua status transport

Description

Displays the status of SIP call service on a SIP gateway.

Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP

TCP or UDP socket.

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session through sgcp tse payload session transport (H.323 voice-service)

session transport (H.323 voice-service)

To configure the underlying transport layer protocol for H.323 messages to be used across all VoIP dial peers, use the session transport command in H.323 voice service configuration mode. To reset the default value, use the no form of this command.

session transport {udp| tcp [calls-per-connection value]}

no session transport

Syntax Description udp tcp calls-per-connection

value

Configures the H.323 dial peer to use the UDP transport layer protocol.

Configures the H.323 dial peer to use the TCP transport layer protocol. This is the default.

Configures the number of calls multiplexed into a single TCP connection.

The number of calls. The range is from 1 to 9999.

The default is 5.

Command Default

TCP is the default session transport protocol; the default calls-per-connection value is 5.

Command Modes

H.323 voice-service configuration (conf-serv-h323)

Command History

Release

12.2(1)T

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced for session initiation protocol (SIP) dial peers.

This command was modified to include support for H323 dial peers and to include the calls-per-connection keyword.

This command was integrated into Cisco IOS Release 12.2(4)T.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

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Examples

The following example shows a dial peer configured to use the UDP transport layer protocol.

Router(conf-voi-serv)#

h323

Router(conf-serv-h323)#

session transport udp

Related Commands

Command h323

Description

Enables H.323 voice service configuration commands.

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session through sgcp tse payload session transport (SIP)

session transport (SIP)

To configure the underlying transport layer protocol for SIP messages to TCP, transport layer security over

TCP (TLS over TCP), or User Datagram Protocol (UDP), use the session transport command in SIP configuration mode. To reset the value of this command to the default, use the no form of this command.

session transport {udp| tcp [tls]}

no session transport {udp| tcp [tls]}

Syntax Description udp tcp tls

Configure SIP messages to use the UDP transport layer protocol. This is the default.

Configure SIP messages to use the TCP transport layer protocol.

(Optional) Configure SIP messages to use the TLS over TCP transport layer protocol.

Command Default

The default for the command is UDP.

Command Modes

Voice service SIP configuration (conf-serv-sip)

Command History

Release

12.2(2)XB

12.2(2)XB2

12.2(8)T

12.2(11)T

12.4(6)T

Modification

This command was introduced in SIP configuration mode.

This command was implemented on the Cisco AS5850 platform.

This command was integrated into Cisco IOS Release 12.2(8)T and support was added for the Cisco 3700 series. Cisco AS5300, Cisco AS5350, Cisco

AS5850, and Cisco AS5400 platforms were not supported in this release.

Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and

Cisco AS5850 platforms.

The optional tls keyword was added to the command.

Usage Guidelines

Use the show sip-ua status command to verify that the transport protocol set with the session

transportcommand matches the protocol set using the transport command in SIP user agent configuration mode.

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Examples

The following example configures the underlying transport layer protocol for SIP messages to UDP:

voice service voip sip session transport udp

The following example configures the underlying transport layer protocol for SIP messages to TCP:

voice service voip sip session transport tcp

The following example configures the underlying transport layer protocol for SIP messages to TLS over TCP:

voice service voip sip session transport tcp tls

Related Commands

Command show sip-ua status transport

Description

Displays the status of SIP call service on a SIP gateway.

Configures the SIP gateway for SIP signaling messages on inbound calls through the SIP TCP or

UDP socket.

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session through sgcp tse payload session-set

session-set

To create a Signlaing System 7 ( SS7)-link-to-SS7-session-set association or to associate an SS7 link with an

SS7 session set on the Cisco 2600-based Signaling Link Terminal (SLT), enter the session-set command in global configuration mode. To remove the link from its current SS7 session set and to add it to SS7 session set 0 (the default), use the no form of this command.

session-set session-set-id

no session-set

Syntax Description

session -set-id SS7 session ID. Valid values are 0 and 1. Default is

0.

Command Default

SS7 session set 0

Command Modes

Global configuration (config)

Command History

Release

12.2(15)T

Modification

This command was introduced on the Cisco 2600-based SLT.

Usage Guidelines

On Cisco AS5350 and Cisco AS5400 platforms, the channel-id command is used to create an

SS7-link-to-SS7-session-set association on the Cisco SLT. The Cisco 26xx platforms do not support the

channel-id command, so channel IDs on the Cisco 26xx-based SLT are implicitly assigned on the basis of the slot location of the WAN interface card (WIC) and the channel group ID used to create the SS7 link.

If this command is omitted, the link is implicitly added to the SS7 session set 0, which is the default.

Examples

The following example shows how the session-set command is used to add the associated SS7 link to an SS7 session set: session-set 1

The following example shows how the no session-set command is used to remove the link from its current

SS7 session set and add it to SS7 session set 0, which is the default: no session-set

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Related Commands

Command channel-id session-set

Description

Assigns a session channel ID to a Signaling System

7 (SS7) serial link or assign an SS7 link to an SS7 session set on a Cisco AS5350 or Cisco AS5400.

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session through sgcp tse payload session-timeout

session-timeout

To specify the maximum amount of time for which a TFTP session can remain open, use the session-timeout command in phone proxy configuration mode. To remove the timeout period of a TFTP session, use the no form of the command.

session-timeout seconds

no session-timeout

Syntax Description

seconds

Maximum length of a TFTP session in seconds. The range is from 60 to 6000. The default is 180 seconds.

Command Default

The session timeout is 180 seconds.

Command Modes

Phone proxy configuration mode (config-phone-proxy)

Command History

Release

15.3(3)M

Modification

This command was introduced.

Usage Guidelines

Examples

The following example shows how to specify a timeout period for a TFTP session of 200 seconds:

Device(config)# voice-phone-proxy first-pp

Device(config-phone-proxy)# session-timeout 200

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set

To create a fault-tolerant or nonfault-tolerant session set with the client or server option, use the set command in backhaul session-manager configuration mode. To delete the set, use the no form of this command.

set set-name {client| server} {ft| nft}

no set set-name {client| server} {ft| nft}

Syntax Description

set -name

client server ft nft

Session-set name.

The session set operates as a client. Select this option for signaling backhaul.

The session set operates as a server.

Fault-tolerant operation. Select fault-tolerant if this session set can contain more than one session group, with each session group connecting the gateway to a different Cisco VSC3000. Fault-tolerance allows the system to operate properly if a session group in the session set fails.

Non-fault-tolerant operation. Select non-fault-tolerant if this session set contains only one session group

(which connects the gateway to a single Cisco

VSC3000).

Command Default

No default behavior or values

Command Modes

Backhaul session manager configuration (config-bsm)

Command History

Release

12.1(1)T

12.2(4)T

12.2(2)XB

12.2(2)XB1

Modification

This command was introduced.

This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

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session through sgcp tse payload set

Release

12.2(8)T

12.2(11)T

Modification

This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for on the Cisco AS5350,

Cisco AS5400, and Cisco AS5850 is not included in this release.

This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco

AS5850 in this release.

Usage Guidelines

Multiple session groups can be associated with a session set.

For signaling backhaul, session sets should be configured to operate as clients.

A session set cannot be deleted unless all session groups associated with the session set are deleted first.

Examples

The following example sets the client set named "set1" as fault-tolerant:

Router(config-bsm)#

set set1 client ft

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set http client cache stale

To set the status of all entries in the HTTP client cache to stale, use the set http client cache stale command in global configuration mode.

set http client cache stale

Syntax Description

This command has no arguments or keywords.

Command Default

Entries in the HTTP client cache are not marked stale manually.

Command Modes

Global configuration (config)

Command History

Release

12.4(15)T

12.4(20)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

Use this command to force the HTTP client to check with the server to see if an updated version of the file exists when any cached entries are requested by the VoiceXML application. If the router is in nonstreaming mode, a conditional reload is sent to the HTTP server. If the router is in streaming mode, an unconditional reload is sent for the refresh. Regardless of which mode the router is in, the VoiceXML application is guaranteed to receive the most up-to-date file when you use the set http client cache stale command.

The show http client cache command shows a pound sign (#) next to the age of entries that are marked stale manually.

Examples

The following example sets the status of all entries in the HTTP client cache to stale:

Router#

set http client cache stale

Related Commands

Command show http client cache

Description

Displays information about the entries contained in the HTTP client cache.

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session through sgcp tse payload set pstn-cause

set pstn-cause

To map an incoming PSTN cause code to a Session Initiation Protocol (SIP) error status code, use the set

pstn-causecommand in SIP user-agent configuration mode. To reset to the default, use the no form of this command.

set pstn-cause value sip-status value

no set pstn-cause

Syntax Description

pstn -cause value

sip -status value

PSTN cause code. Range is from 1 to 127

SIP status code that is to correspond with the PSTN cause code. Range is from 400 to 699.

Command Default

18

19

20

3

17

21

22

26

27

28

The default mappings defined in the following table are used:

Table 5: Default PSTN Cause Codes Mapped to SIP Events

PSTN Cause Code

1

2

Description

Unallocated number

No route to specified transit network

No route to destination

User busy

No user responding

No answer from the user

Subscriber absent

Call rejected

Number changed

Non-selected user clearing

Destination out of order

Address incomplete

SIP Event

404 Not found

404 Not found

404 Not found

486 Busy here

480 Temporarily unavailable

403 Forbidden

410 Gone

404 Not found

404 Not found

484 Address incomplete

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34

38

41

PSTN Cause Code

29

31

Description

Facility rejected

Normal, unspecified

No circuit available

Network out of order

Temporary failure

SIP Event

501 Not implemented

404 Not found

503 Service unavailable

503 Service unavailable

503 Service unavailable

42

47

55

Switching equipment congestion 503 Service unavailable

Resource unavailable 503 Service unavailable

Incoming class barred within the

Closed User Group (CUG)

403 Forbidden

Bearer capability not authorized 403 Forbidden 57

58

65

79

87

88

95

102

Bearer capability not currently available

501 Not implemented

Bearer capability not implemented 501 Not implemented

Service or option not implemented 501 Not implemented

503 Service unavailable User not member of the Closed

User Group (CUG)

Incompatible destination

Invalid message

Recover on Expires timeout

400 Bad request

400 Bad request

408 Request timeout

400 Bad request 111

Any code other than those listed above

Protocol error

500 Internal server error

Command Modes

SIP UA configuration (config-sip-ua)

Command History

Release

12.2(2)XB

Modification

This command was introduced.

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session through sgcp tse payload set pstn-cause

Release

12.2(2)XB2

12.2(8)T

Modification

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support for on the Cisco AS5300 Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

Usage Guidelines

A PSTN cause code can be mapped only to one SIP status code at a time.

Examples

The following example maps a SIP status code to correspond to a PSTN cause code:

Router(config)#

sip-ua

Router(config-sip-ua)#

set pstn-cause 111 sip-status 400

Router(config-sip-ua)#

exit

Related Commands

Command

set sip -status

Description

Sets an incoming SIP error status code to a PSTN release cause code.

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set sip-status

To map an incoming Session Initiation Protocol (SIP) error status code to a PSTN cause code, use the set

sip-statuscommand in SIP user-agent configuration mode. To reset to the default, use the no form of this command.

set sip-status value pstn-cause value

no set sip-status

Syntax Description

sip -status value

pstn -cause value

SIP status code. Range is from 400 to 699.

PSTN cause code that is to correspond with the SIP status code. Range is from 1 to 127.

Command Default

The default mappings defined in the table below are used:

Table 6: Default SIP Events Mapped to PSTN Cause Codes

SIP Event

400 Bad request

401 Unauthorized

402 Payment required

403 Forbidden

404 Not found

405 Method not allowed

406 Not acceptable

407 Proxy authentication required 21

408 Request timeout 102

409 Conflict

410 Gone

41

1

21

57

1

127

PSTN Cause Code

127

57

Description

Interworking, unspecified

Bearer capability not authorized

Call rejected

Bearer capability not authorized

Unallocated number

Interworking, unspecified

Call rejected

Recover on Expires timeout

Temporary failure

Unallocated number

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set sip-status

SIP Event

411 Length required

413 Request entity too long

480 Temporarily unavailable

481 Call leg does not exist

482 Loop detected

483 Too many hops

PSTN Cause Code

127

414 Request URI (URL) too long

415 Unsupported media type 79

420 Bad extension 127

18

127

484 Address incomplete

485 Address ambiguous

486 Busy here

487 Request canceled

488 Not acceptable here

500 Internal server error

501 Not implemented

41

79

127

127

28

1

17

502 Bad gateway

503 Service unavailable

504 Gateway timeout

505 Version not implemented

580 Precondition failed

600 Busy everywhere

603 Decline

604 Does not exist anywhere

38

63

102

127

47

17

21

1

session through sgcp tse payload

Description

Interworking, unspecified

Service or option not available

Interworking, unspecified

No user response

Interworking, unspecified

Address incomplete

Unallocated number

User busy

Interworking, unspecified

Interworking, unspecified

Temporary failure

Service or option not implemented

Network out of order

Service or option unavailable

Recover on Expires timeout

Interworking, unspecified

Resource unavailable, unspecified

User busy

Call rejected

Unallocated number

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SIP Event

606 Not acceptable

PSTN Cause Code

58

Description

Bearer capability not currently available

Command Modes

SIP UA configuration (config-sip-ua)

Command History

Release

12.2(2)XB

12.2(2)XB2

12.2(8)T

Modification

This command was introduced.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

Usage Guidelines

A SIP status code can be mapped to many PSTN cause codes. For example, 503 can be mapped to 34, 38, and 58.

Examples

The following example maps a PSTN cause code to correspond to a SIP status code:

Router(config)#

sip-ua

Router(config-sip-ua)#

set sip-status 400 pstn-cause 16

Related Commands

Command

set pstn -cause

Description

Sets an incoming PSTN cause code to a SIP error status code.

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session through sgcp tse payload settle-call

settle-call

To force a call to be authorized with a settlement server that uses the address resolution method specified in the session target command, use the settle-callcommand in dial-peer configuration mode. To ensure that no authorization is performed by a settlement server, use the no form of this command.

settle-call provider-number

no settle-call provider-number

Syntax Description

provider -number Digit defining the ID of a particular settlement server.

The only valid entry is 0.

Note

If session target type is settlement, the

provider-number argument in the session

target and settle-call commands should be identical.

Command Default

No default behavior or values.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Release

12.1(1)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.

Usage Guidelines

With the session target command, a dial peer can determine the address of the terminating gateway through the ipv4, dns, ras, and settlement keywords.

If the session target is not settlement, and the settle-call provider-number argument is set, the gateway resolves address of the terminating gateway using the specified method and then requests the settlement server to authorize that address and create a settlement token for that particular address. If the server cannot authorize the terminating gateway address suggested by the gateway, the call fails.

Do not combine the session target types ras and settle-call. Combination of session target types is not supported.

Examples

The following example sets a call to be authorized with a settlement server that uses the address resolution method specified in the session target: dial-peer voice 10 voip destination-pattern 1408.......

session target ipv4:172.22.95.14

settle-call 0

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Related Commands

Command session target settle-call

Description

Specifies a network-specific address for a specified dial peer.

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settlement

To enter settlement configuration mode and specify the attributes specific to a settlement provider, use the

settlement command in global configuration mode. To disable the settlement provider, use the no form of this command.

settlement provider-number

no settlement provider-number

Syntax Description

provider -number Digit that defines a particular settlement server. The only valid entry is 0.

Command Default

0

Command Modes

Global configuration (config)

Command History

Release

12.0(4)XH1

12.1(1)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.

This command was integrated into Cisco IOS Release 12.1(1)T.

Usage Guidelines

The variable provider-number defines a particular settlement provider. For Cisco IOS Release 12.1, only one clearinghouse per system is allowed, and the only valid value for provider-number is 0.

Examples

This example enters settlement configuration mode: settlement 0

Related Commands

Command

connection -timeout

customer -id

Description

Configures the length of time for which a connection is maintained after a communication exchange is completed.

Identifies a carrier or ISP with a settlement provider.

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Command

device -id

encryption

max -connection

response -timeout

retry -delay

retry -limit

session -timeout

show settlement shutdown type settlement

Description

Specifies a gateway associated with a settlement provider.

Sets the encryption method to be negotiated with the provider.

Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.

Configures the maximum time to wait for a response from a server.

Sets the time between attempts to connect with the settlement provider.

Sets the connection retry limit.

Sets the interval for closing the connection when there is no input or output traffic.

Displays the configuration for all settlement server transactions.

Brings up the settlement provider.

Configures an SAA-RTR operation type.

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session through sgcp tse payload settlement roam-pattern

settlement roam-pattern

To configure a pattern that must be matched to determine if a user is roaming, use the settlement

roam-patterncommand in global configuration mode. To delete a particular pattern, use the no form of this command.

settlement provider-number roam-pattern pattern {roaming| noroaming}

no settlement provider-number roam-pattern pattern {roaming| noroaming}

Syntax Description

provider -number

pattern

roaming noroaming

Digit defining the ID of particular settlement server.

The only valid entry is 0.

User account pattern.

Specifies that a user is roaming.

Specifies that a user is not roaming.

Command Default

No default pattern is configured.

Command Modes

Global configuration (config)

Command History

Release

12.1(1)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.

Usage Guidelines

Multiple roam patterns can be entered on one gateway.

Examples

The following example shows how to configure a pattern that determines if a user is roaming: settlement 0 roam-pattern 1222 roaming settlement 0 roam-pattern 1333 noroaming settlement 0 roam-pattern 1444 roaming settlement 0 roam-pattern 1555 noroaming

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Related Commands

Command roaming (settlement) settlement settlement roam-pattern

Description

Enables the roaming capability for a settlement provider.

Enters settlement configuration mode.

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session through sgcp tse payload sgcp

sgcp

To start and allocate resources for the Simple Gateway Control Protocol (SGCP) daemon, use the sgcp command in global configuration mode. To terminate all calls, release all allocated resources, and kill the

SGCP daemon, use the no form of this command.

sgcp no sgcp

Syntax Description

This command has no arguments or keywords.

Command Default

The SGCP daemon is not enabled.

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300 only and was not generally available.

This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and Cisco MC3810.

Usage Guidelines

When the SGCP daemon is not active, all SGCP messages are ignored.

When you enter the no sgcp command, the SGCP process is removed.

Note

After you enter the no sgcp command, you must save the configuration and reboot the router for the disabling of SGCP to take effect.

Examples

The following example enables the SGCP daemon: sgcp

The following example disables the SGCP daemon: no sgcp

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Related Commands

Command sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp

Description

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp call-agent

sgcp call-agent

To define the IP address of the default Simple Gateway Control Protocol (SGCP) call agent in the router configuration file, use the sgcp call-agent command in global configuration mode. To remove the IP address of the default SGCP call agent from the router configuration, use the no form of this command.

sgcp call-agent ipaddress [: udp port]

no sgcp call-agent ipaddress

Syntax Description

ipaddress

:udp port

IP address or hostname of the call agent.

(Optional) UDP port of the call agent.

Command Default

No IP address is configured.

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300 only and was not generally available.

This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and Cisco MC3810.

Usage Guidelines

This command defines the IP address of the default SGCP call agent to which the router sends an initial RSIP

(Restart In Progress) packet when the router boots up. This is used for initial bootup only before the SGCP call agent contacts the router acting as the gateway.

When you enter the no sgcp call-agent command, only the IP address of the default SGCP call agent is removed.

Examples

The following example enables SGCP and specifies the IP address of the call agent: sgcp sgcp call-agent 209.165.200.225

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Related Commands

Command sgcp sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp call-agent

Description

Starts and allocates resources for the SGCP daemon.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp graceful-shutdown

sgcp graceful-shutdown

To block all new calls and gracefully terminate all existing calls (wait for the caller to end the call), use the

sgcp graceful-shutdowncommand in global configuration mode. To unblock all calls and allow new calls to go through, use the no form of this command.

sgcp graceful-shutdown no sgcp graceful-shutdown

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values.

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300 and was not generally available.

This command was implemented on the Cisco MC3810 and Cisco 3600 series

(except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and Cisco MC3810.

Usage Guidelines

Once you issue this command, all requests for new connections (CreateConnection requests) are denied. All existing calls are maintained until users terminate them, or until you enter the no sgcpcommand. When the last active call is terminated, the SGCP daemon is terminated, and all resources allocated to it are released.

Examples

The following example blocks all new calls and terminates existing calls: sgcp graceful-shutdown

Related Commands

Command sgcp sgcp call-agent sgcp max-waiting-delay

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

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Command sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp graceful-shutdown

Description

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp max-waiting-delay

sgcp max-waiting-delay

To set the Simple Gateway Control Protocol (SGCP) maximum waiting delay to prevent restart avalanches, use the sgcp max-waiting-delaycommand in global configuration mode. To reset to the default, use the no form of this command.

sgcp max-waiting-delay delay

no sgcp max-waiting-delay delay

Syntax Description

delay

Maximum waiting delay (MWD), in milliseconds.

Range is from 0 to 600000. Default is 3000.

Command Default

3,000 ms

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300, and was not generally available.

This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Examples

The following example sets the maximum wait delay value to 40 ms: sgcp max-waiting-delay 40

Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

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Command sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp max-waiting-delay

Description

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp modem passthru

sgcp modem passthru

To enable Simple Gateway Control Protocol (SGCP) modem or fax pass-through, use the sgcp modem

passthru command in global configuration mode. To disable SGCP modem or fax pass-through, use the no form of this command.

sgcp modem passthru {ca| cisco| nse}

no sgcp modem passthru {ca| cisco| nse}

Syntax Description ca cisco nse

Call-agent-controlled modem upspeed-method violation message.

Cisco-proprietary upspeed method based on the protocol.

NSE-based modem upspeed method.

Command Default

SGCP modem or fax pass-through is disabled by default.

Command Modes

Global configuration (config)

Command History

Release

12.0(7)XK

12.1(2)T

Modification

This command was introduced on the Cisco MC3810 and the Cisco 3600 series

(except the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

You can use this command for fax pass-through because the answer tone can come from either modem or fax transmissions. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions.

If you use the nse option, you must also configure the sgcp tse payload command.

Examples

The following example configures SGCP modem pass-through using the call-agent upspeed method: sgcp modem passthru ca

The following example configures SGCP modem pass-through using the proprietary Cisco upspeed method: sgcp modem passthru cisco

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The following example configures SGCP modem pass-through using the NSE-based modem upspeed: sgcp modem passthru nse sgcp tse payload 110

Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp quarantine-buffer disable

sgcp quarantine-buffer disable

To disable the Simple Gateway Control Protocol (SGCP) quarantine buffer, use the sgcp quarantine-buffer

disablecommand in global configuration mode. To reenable the SGCP quarantine buffer, use the no form of this command.

sgcp quarantine-buffer disable no sgcp quarantine-buffer disable

Syntax Description

This command has no arguments or keywords.

Command Default

The SGCP quarantine buffer is enabled.

Command Modes

Global configuration (config)

Command History

Release

12.0(7)XK

12.1(2)T

Modification

This command was introduced on the Cisco MC3810 and the Cisco 3600 series

(except for the Cisco 3620) in a private release that was not generally available.

This command was on the Cisco 3600 series and the Cisco MC3810.

Usage Guidelines

The SGCP quarantine buffer is the mechanism for buffering the SGCP events between two notification-request

(RQNT) messages.

Examples

The following example disables the SGCP quarantine buffer: sgcp quarantine-buffer disable

Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

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Command sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp quarantine-buffer disable

Description

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp request retries

sgcp request retries

To specify the number of times to retry sending notify and delete messages to the Simple Gateway Control

Protocol (SGCP) call agent, use the sgcp request retriescommand in global configuration mode. To reset to the default, use the no form of this command.

sgcp request retries count

no sgcp request retries

Syntax Description

count

Number of times that a notify and delete message is retransmitted to the SGCP call agent before it is dropped. Range is from 1 to 100. Default is 3.

Command Default

3 times

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300 and was not generally available.

This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

The actual retry count may be different from the value you enter for this command. The retry count is also limited by the call agent. If there is no response from the call agent after 30 seconds, the gateway does not retry anymore, even though the number set using the sgcp request retriescommand has not been reached.

The router stops sending retries after 30 seconds, regardless of the setting for this command.

Examples

The following example configures the system to send the sgcp command 10 times before dropping the request: sgcp request retries 10

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Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp request retries

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp request timeout

sgcp request timeout

To specify how long the system should wait for a response to a request, use the sgcp request timeoutcommand in global configuration mode. To reset to the default, use the no form of this command.

sgcp request timeout timeout

no sgcp request timeout

Syntax Description

timeout

Time to wait for a response to a request, in milliseconds. Range is from 1 to 10000. Default is

500.

Command Default

500 ms

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300 and was not generally available.

This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

This command is used for "notify" and "delete" messages, which are sent to the SGCP call agent.

Examples

The following example configures the system to wait 40 ms for a reply to a request: sgcp request timeout 40

Related Commands

Command sgcp sgcp call-agent

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

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Command sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload sgcp request timeout

Description

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp restart

sgcp restart

To trigger the router to send a Restart in Progress (RSIP) message to the Simple Gateway Control Protocol

(SGCP) call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller, use the sgcp restartcommand in global configuration mode. To reset to the default, use the

no form of this command.

sgcp restart {delay delay| notify}

no sgcp restart {delay delay| notify}

Syntax Description

delay delay

notify

Restart delay, in milliseconds. Range is from 0 to

600. Default is 0.

Restarts notification upon the SGCP/digital interface state transition.

Command Default

0 ms

Command Modes

Global configuration(config)

Command History

Release

12.0(7)XK

12.1(2)T

Modification

This command was introduced on the Cisco MC3810 and the Cisco 3600 series

(except the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

Use this command to send RSIP messages from the router to the SGCP call agent. RSIP messages are used to synchronize the router and the call agent. RSIP messages are also sent when the sgcp command is entered to enable the SGCP daemon.

You must enter the notify option to enable RSIP messages to be sent.

Examples

The following example configures the system to wait 40 ms before restarting SGCP: sgcp restart delay 40

The following example configures the system to send an RSIP notification to the SGCP call agent when the

T1 controller state changes: sgcp restart notify

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Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp retransmit timer sgcp timer sgcp tse payload sgcp restart

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Configures the SGCP retransmission timer to use a random algorithm method.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp retransmit timer

sgcp retransmit timer

To configure the Simple Gateway Control Protocol (SGCP) retransmission timer to use a random algorithm, use the sgcp retransmit timercommand in global configuration mode. To reset to the default, use the no form of this command.

sgcp retransmit timer random no sgcp retransmit timer random

Syntax Description random

SGCP retransmission timer uses a random algorithm.

Command Default

The SGCP retransmission timer does not use a random algorithm.

Command Modes

Global configuration (config)

Command History

Release

12.0(7)XK

12.1(2)T

Modification

This command was introduced on the Cisco 3600 series and the Cisco MC3810 in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

Use this command to enable the random algorithm component of the retransmission timer. For example, if the retransmission timer is set to 200 ms, the first retransmission timer is 200 ms, but the second retransmission timer picks up a timer value randomly between either 200 or 400. The third retransmission timer picks up a timer value randomly of 200, 400, or 800 as shown below:

• First retransmission timer: 200

• Second retransmission timer: 200 or 400

• Third retransmission timer: 200, 400, or 800

• Fourth retransmission timer: 200, 400, 800, or 1600

• Fifth retransmission timer: 200, 400, 800, 1600, or 3200 and so on.

After 30 seconds, the retransmission timer no longer retries.

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Examples

The following example sets the retransmission timer to use a random algorithm: sgcp retransmit timer random

Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp timer sgcp tse payload

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures how the gateway detects the RTP stream host.

Enables Inband TSE for fax/modem operation.

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session through sgcp tse payload sgcp timer

sgcp timer

To configure how the gateway detects the Real-Time Transport Protocol (RTP) stream lost, use the sgcp

timercommand in global configuration mode. To reset to the default, use the no form of this command.

sgcp timer {receive-rtcp timer| rtp-nse timer}

no sgcp timer {receive-rtcp timer| rtp-nse timer}

Syntax Description

receive -rtcp timer

rtp -nse timer

RTP Control Protocol (RTCP) transmission interval, in milliseconds. Range is from 1 to 100. Default is 5.

RTP named signaling event (NSE) timeout, in milliseconds. Range is from 100 to 3000. Default is

200.

Command Default

receive -rtcp: 5 ms

rtp -nse: 200 ms

Command Modes

Global configuration (config)

Command History

Release

12.0(5)T

12.0(7)XK

12.1(2)T

Modification

This command was introduced in a private release on the Cisco AS5300 and was not generally available.

This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

The RTP NSE timer is used for proxy ringing (the ringback tone is provided at the originating gateway).

Examples

The following example sets the RTPCP transmission interval to 100 ms: sgcp timer receive-rtcp 100

The following example sets the NSE timeout to 1000 ms: sgcp timer rtp-nse 1000

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Related Commands

Command sgcp sgcp call-agent sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp tse payload sgcp timer

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.

Enables Inband TSE for fax/modem operation.

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sgcp tse payload

To enable Inband Telephony Signaling Events (TSE) for fax and modem operation, use the sgcp tse

payloadcommand in global configuration mode. To reset to the default, use the no form of this command.

sgcp tse payload type

no sgcp tse payload type

Syntax Description

type

TSE payload type. Range is from 96 to 119. Default is 0, meaning that the command is disabled.

Command Default

0 (disabled)

Command Modes

Global configuration(config)

Command History

Release

12.0(7)XK

12.1(2)T

Modification

This command was introduced on the Cisco MC3810 and the Cisco 3600 series

(except the Cisco 3620) in a private release that was not generally available.

This command was implemented on the Cisco 3600 series and the Cisco

MC3810.

Usage Guidelines

Because this command is disabled by default, you must specify a TSE payload type.

If you set the sgcp modem passthru command to the nse value, then you must configure this command.

Examples

The following example sets Simple Gateway Control Protocol (SGCP) modem pass-through using the

NSE-based modem upspeed and the Inband Telephony Signaling Events payload value set to 110: sgcp modem passthru nse sgcp tse payload 110

Related Commands

Command sgcp sgcp call-agent

Description

Starts and allocates resources for the SGCP daemon.

Defines the IP address of the default SGCP call agent.

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Command sgcp graceful-shutdown sgcp max-waiting-delay sgcp modem passthru sgcp quarantine-buffer disable sgcp request retries sgcp request timeout sgcp restart sgcp retransmit timer sgcp timer sgcp tse payload

Description

Gracefully terminates all SGCP activity.

Sets the SGCP maximum waiting delay to prevent restart avalanches.

Enables SGCP modem or fax pass-through.

Disables the SGCP quarantine buffer.

Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.

Specifies how long the system should wait for a response to a request.

Triggers the router to send an RSIP message to the

SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.

Configures the SGCP retransmission timer to use a random algorithm method.up or down so that the call agent can synchronize

Configures how the gateway detects the RTP stream host.

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source filter

To filter Real-time Transport Protocol (RTP) packets with a source IP address and port number that are different from the one negotiated through Session Initiation Protocol (SIP) signaling, use the source filter command in voice service SIP configuration mode. To disable filtering, use the no form of this command.

source filter no source filter

Command Default

RTP source filtering is disabled.

Command Modes

Voice service SIP configuration (conf-serv-sip)

Command History

Release

12.4(6)T

Modification

This command was introduced.

Usage Guidelines

Public Switched Telephone Network (PSTN) callers may experience crosstalk when the SIP IOS gateway receives an invalid RTP stream destined to the same IP address and port of an active call. The invalid stream has a different source IP address and port than the one negotiated using SIP Session Description Protocol

(SDP). The Digital Signal Processor (DSP) within the gateway mixes both the valid and invalid RTP streams and plays it to the PSTN caller. Use the source filter command when you want to filter RTP packets with a source IP address and port number that are different from the one negotiated through SIP signaling.

Examples

The following example shows how to filter RTP packets:

Device>

enable

Device#

configure terminal

Device(config)#

voice service voip

Device(conf-voi-serv)#

sip

Device(conf-serv-sip)#

source filter

Related Commands

Command sip voice service voip

Description

Enters SIP configuration mode.

Specifies the voice-encapsulation type and enters voice service configuration mode.

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show aal2 profile, page 193

show atm video-voice address, page 196

show auto-config, page 198

show backhaul-session-manager group, page 200

show backhaul-session-manager session, page 203

show backhaul-session-manager set, page 206

show call accounting-template voice, page 208

show call active fax, page 212

show call active media, page 223

show call active total-calls, page 231

show call active video, page 232

show call active voice, page 243

show call application app-level, page 266

show call application gateway-level, page 270

show call application interface, page 273

show call application services registry, page 277

show call application session-level, page 279

show call application sessions, page 291

show call application voice, page 295

show call fallback cache, page 301

show call fallback config, page 303

show call fallback stats, page 305

show call filter components, page 307

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show call filter match-list, page 309

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show aal2 profile

To display the ATM adaptation layer 2 (AAL2) profiles configured on the system, use the show aal2 profile command in privileged EXEC mode.

show aal2 profile all {itut profile-number| atmf profile-number| custom profile-number}

Syntax Description all itut atmf custom

profile -number

Displays ITU-T, ATMF, and custom AAL2 profiles configured on the system.

Displays ITU-T profiles configured on the system.

Displays ATMF profiles configured on the system.

Displays custom profiles configured on the system.

AAL2 profile number to display. Choices are as follows:

For ITU-T:

• 1 = G.711 u-law

• 2 = G.711 u-law with silence insertion descriptor (SID)

• 7 = G.711 u-law and G.729ar8

For ATMF: None. ATMF is not supported.

For custom:

• 100 = G.711 u-law and G.726r32

• 110 = G.711 u-law, G.726r32, and G.729ar8

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(1)XA

12.1(2)T

12.2(2)T

Modification

This command was introduced on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was implemented on the Cisco 7200 series.

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Usage Guidelines

This command applies to AAL2 VoATM applications on the Cisco 7200 series routers.

Examples

The following command displays all of the configured profiles in the system:

Router#

show aal2 profile all

Printing all the Profiles in the system

Profile Type: ITUT Profile Number: 1 SID Support: 0

Red enable: 1 Num entries: 1

Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15

Profile Type: ITUT Profile Number: 2 SID Support: 1

Red enable: 1 Num entries: 1

Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15

Profile Type: custom Profile Number: 100 SID Support: 1

Red enable: 1 Num entries: 2

Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7

Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15

Profile Type: ITUT Profile Number: 7 SID Support: 1

Red enable: 1 Num entries: 2

Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15

Coding type: g729ar8 Packet length: 10 UUI min: 0 UUI max: 15

Profile Type: custom Profile Number: 110 SID Support: 1

Red enable: 1 Num entries: 3

Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7

Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15

Coding type: g729ar8 Packet length: 30 UUI min: 8 UUI max: 15

The table below describes significant fields shown in this output.

Table 7: show aal2 profile all Field Descriptions

Field

Coding type

ITUT Profile Number

Num entries

Packet length

Profile Type

Red enable

SID Support

UUI max

UUI min

Description

Voice compression algorithm.

Predefined combination of one or more codec types configured for a digital signal processor (DSP).

Number of profile elements.

Sample size.

Category of codec types configured on DSP. Possible types are ITU-T, ATMF, and custom.

Redundancy for type 3 packets.

Silence insertion descriptor.

Maximum sequence number on the voice packets.

Minimum sequence number on the voice packets.

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Related Commands

Command codec aal2-profile show aal2 profile

Description

Sets the codec profile for a DSP on a per-call basis.

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show atm video-voice address

To display the network service access point (NSAP) address for the ATM interface, enter the show atm

video-voice addresscommand inprivileged EXEC mode.

show atm video-voice address

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)XK

12.0(7)T

Modification

This command was introduced on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.0(7)T.

Usage Guidelines

Use this command to review ATM interface NSAP addresses that have been assigned with the atm video aesa command and to ensure that ATM management is confirmed for those addresses.

Examples

The following example displays ATM interface NSAP addresses:

Router#

show atm video-voice address

nsap address type ilmi status

47.0091810000000002F26D4901.00107B4832E1.FE

VOICE_AAL5 Confirmed

47.0091810000000002F26D4901.00107B4832E1.C8

VIDEO_AAL1 Confirmed

The table below describes the significant fields shown in the output.

Table 8: show atm video-voice address Field Descriptions

Field

NSAP address

Type

ILMI status

Description

NSAP address for the ATM interface.

Type of ATM interface.

Integrated Local management Interface (ILMI) protocol status for the ATM interface.

Related Commands

Command codec aal2-profile

Description

Sets the codec profile for a DSP on a per-call basis.

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show auto-config

To display the current status of auto-configuration applications, use the show auto-configcommand in privileged EXEC mode.

show auto-config [application sccp]

Syntax Description application sccp

Displays the current status of only the Skinny Client

Control Protocol (SCCP) application.

Command Modes

Privileged EXEC (#)

Command History

Examples

Release

12.3(8)XY

12.3(14)T

Modification

This command was introduced on the Communication Media Module.

This command was integrated into Cisco IOS Release 12.3(14)T.

The following is sample output from show auto-config command:

Router#

show auto-config application sccp

auto-config application: sccp auto-config admin state: ENABLED & ACTIVE download retries: (3) download timeout: no timeout, continuous retry server(s): 172.19.240.41 172.19.240.40 172.19.240.42

Configuration Download statistics:

Download Attempted : 2

Download Successful

Download Failed

Configuration Attempted

Configuration Successful

: 2

: 0

: 2

: 2

Configuration Failed(parsing): 0

Configuration Failed(config) : 0

Configuration Error History:

The table below describes the significant fields shown in the display.

Table 9: show auto-config Field Descriptions

Field

ENABLED

ACTIVE

Description

Shows auto-config application: SCCP is enabled.

Shows the SCCP application has registered to use auto-configuration.

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Field

timeout

Related Commands

Command auto-config debug auto-config debug sccp config show auto-config

Description

Shows timeout is set to 0, continuous retry without timeout.

Description

Enables auto-configuration or enters auto-config application configuration mode for the SCCP application.

Enables debugging for auto-configuration applications.

Enables SCCP event debugging.

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show backhaul-session-manager group

To display the status, statistics, or configuration for a particular session group or all available session groups, use the show backhaul-session-manager groupcommand in privileged EXEC mode.

show backhaul-session-manager group {status| stats| cfg} {all| name group-name}

Syntax Description status stats cfg all name

group -name

Status for available session groups.

Statistics for available session groups.

Configuration for available session groups.

Specified parameters for all session groups.

A particular session group.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(1)T

12.2(2)T

12.2(4)T

12.2(2)XB

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was introduced on the Cisco AS5300.

This command was implemented on the Cisco 7200 series.

This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for the Cisco AS5300,

Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

This command is supported on the Cisco AS5300, Cisco AS5350, Cisco

AS5400, and Cisco AS5850 in this release.

Examples

The following example displays statistics for all session groups:

Router# s

how backhaul-session-manager group stats all

Session-Group grp1 statistics

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Successful Fail-Overs :0

Un-Successful Fail-Over attempts:0

Active Pkts receive count :0

Standby Pkts receive count :0

Total PDUs dispatch err :0

The following example displays the current configuration for all session groups:

Router#

show backhaul-session-manager group cfg all

Session-Group

Group Name :grp1

Set Name :set1

Sessions :3

Dest:10.5.0.3 8304 Local:10.1.2.15 8304 Priority:0

Dest:10.5.0.3 8300 Local:10.1.2.15 8300 Priority:2

Dest:10.5.0.3 8303 Local:10.1.2.15 8303 Priority:2

RUDP Options timer cumulative ack :100 timer keepalive timer retransmit

:1000

:300 timer transfer state :2000 receive max :32 cumulative ack max :3 retrans max :2 out-of-sequence max :3 auto-reset max :5

The following example displays the current state of all session groups. The group named "grp1" belongs to the set named "set1".

Router#

show backhaul-session-manager group status all

Session-Group

Group Name :grp1

Set Name :set1

Status :Group-OutOfService

Status (use) :Group-None

The table below describes the significant fields shown in the output.

Table 10: show backhaul-session-manager group Field Descriptions

Field

RUDP Options

Status

Descrption

Reliable User datagram Protocol (RUDP) options.

One of the following:

• Group-OutOfService--No session in the group has been established.

• Group-Inservice--At least one session in the group has been established.

Status (use) One of the following:

• Group-Standby--The virtual switch controller

(VSC) connected to the other end of this group goes into standby mode.

• Group-Active--The VSC connected to the other end of this group is the active VSC.

• Group-None--The VSC has not yet declared its intent.

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Related Commands

Command show backhaul-session-manager session show backhaul-session-manager set

Description

Displays status, statistics, or configuration of sessions.

Displays session groups associated with a specific session set or all session sets.

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show backhaul-session-manager session

To display various information about a session or sessions, use the show backhaul-session-manager session command in privileged EXEC mode.

show backhaul-session-manager session {all| ip ip-address}

Syntax Description all ip

ip -address

Information is displayed about all available sessions.

Information is displayed about the session associated with this IP address only.

IP address of the local or remote session.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(1)T

12.2(2)T

12.2(4)T

12.2(2)XB

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was introduced on the Cisco AS5300.

This command was implemented on the Cisco 7200 series.

This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for the Cisco AS5350,

Cisco AS5400, and Cisco AS5850 is not included in this release.

This command was implemented on the Cisco AS5350, Cisco AS5400, and

Cisco AS5850.

Examples

The following command displays information for all available sessions:

Router#

show backhaul-session-manager session all

Session information --

Session-id:35

Group:grp1 /*this session belongs to the group named 'grp1' */

Configuration:

Local:10.1.2.15

, port:8303

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Remote:10.5.0.3

Priority:2

, port:8303

RUDP Option:Client, Conn Id:0x2

State:

Status:OPEN_WAIT, Use-status:OOS, /*see explanation below */

Statistics:

# of resets:0

# of auto_resets 0

# of unexpected RUDP transitions (total) 0

# of unexpected RUDP transitions (since last reset) 0

Receive pkts Total:0 , Since Last Reset:0

Recieve failures Total:0 ,Since Last Reset:0

Transmit pkts - Total:0, Since Last Reset:0

Transmit Failures (PDU Only)

Due to Blocking (Not an Error) - Total:0, Since Last Reset:0

Due to causes other than Blocking - Total:0, Since Last

Reset:0

Transmit Failures (NON-PDU Only)

Due to Blocking(Not an Error) - Total:0, Since Last Reset:0

Due to causes other than Blocking - Total:0, Since Last

Reset:0

RUDP statistics

Open failures:0

Not ready failures:0

Conn Not Open failures:0

Send window full failures:0

Resource unavailble failures:0

Enqueue failures:0

The table below describes significant fields shown in this output.

Table 11: show backhaul-session-manager session Field Descriptions

Field

State

Use-status

Description

Can be any of the following:

• OPEN--The connection is established.

• OPEN_WAIT--The connection is awaiting establishment.

• OPEN_XFER--Session failover is in progress for this session, which is a transient state.

• CLOSE--The session is down, also a transient state.

The session waits a fixed amount of time and then moves to OPEN_WAIT.

Indicates whether PRI signaling traffic is currently being transported over this session. Can be either of the following:

• OOS--The session is not being used to transport signaling traffic. Out of service (OOS) does not indicate if the connection is established.

• IS--The session is being used currently to transport all PRI signaling traffic. In service (IS) indicates that the connection is established.

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Related Commands

Command show backhaul-session-manager group show backhaul-session-manager set

Description

Displays status, statistics, or configuration of a specific session group or all session groups.

Displays session groups associated with a specific session set or all session sets.

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show backhaul-session-manager set

To display session groups associated with a specified session set or all session sets, use the show

backhaul-session-manager setcommand in privileged EXEC mode.

show backhaul-session-manager set {all| name session-set-name}

Syntax Description all

name session -set -name

All available session sets.

A specified session set.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(1)T

12.2(2)T

12.2(4)T

12.2(2)XB

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was introduced on the Cisco AS5300.

This command was implemented on the Cisco 7200 series.

This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for the Cisco AS5350,

Cisco AS5400, and Cisco AS5850 is not included in this release.

This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco

AS5850 in this release.

Examples

The following command displays session groups associated with all session sets:

Router#

show backhaul-session-manager set all

Related Commands

Command

show backhaul -session-manager group

Description

Displays status, statistics, or configuration of a specific session group or all session groups.

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Command

show backhaul -session-manager session

Description

Displays status, statistics, or configuration of a session or all sessions.

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show aal2 profile through show call filter match-list show call accounting-template voice

show call accounting-template voice

To display accounting template activity, use the show call accounting-template voice command in privileged

EXEC mode.

show call accounting-template voice [acctTempName| master| qdump| summary]

Syntax Description

acctTempName

master qdump

summary

(Optional) Name of the accounting template.

(Optional) Displays all vendor-specific attributes

(VSAs) that are filtered by accounting templates.

(Optional) Displays template activity in the service and free queues.

(Optional) Lists names of all the accounting templates and the number of attributes in each template currently being used.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(11)T

Modification

This command was introduced on the Cisco 3660, Cisco AS5300, Cisco

AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.

Usage Guidelines

• The show call accounting-template voice command displays the status and attributes defined in each template after it is configured.

• The show call accounting-template voice acctTempName command displays the status of a specific template and the attributes (VSAs) that are defined for that template.

• The show call accounting-template voice master command displays all VSAs that can be filtered by accounting templates.

• The show call accounting-template voice qdump command displays template activity in the service

(svc) and free queues. It displays the template URL, the number of legs on which a template is active, and the state of a template.

• After an accounting template is defined, it is put in the svc queue to serve new incoming calls.

When a running accounting template is undefined or reloaded during an active call, the template is moved from the svc queue to the free queue and can be reused after all the active calls stop

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Examples show call accounting-template voice

referencing it. Templates that are reloaded or undefined and that are referenced during an active call are considered to be in a "dirty" state and are called dirty templates.

• To ensure that start and stop records correspond on an active call that is referencing a dirty template, all dirty templates must be kept alive until all active calls referencing that dirty template are released.

After all active calls are released, the reloaded templates are applied to the next call.

• The show call accounting-template voice summary command displays the current status of all the accounting templates that are configured. It shows if the template was loaded and if it is running successfully.

The following example displays details about two templates named "cdr1" and "cdr2".

Router#

show call accounting-template voice

CDR template cdr1 is running url: tftp://sanjoe/santa/abc/Templates/cdr1.cdr

The last load was successful.

attr: h323-call-origin (56) attr: h323-call-type (57) attr: h323-gw-id (65) attr: subscriber (79) attr: in-portgrp-id (80) attr: out-portgrp-id (81)

Totally 6 attrs defined.

CDR template cdr2 is running url: tftp://sanjoe/santa/abc/Templates/cdr2.cdr

The last load was successful.

attr: h323-call-origin (56) attr: h323-call-type (57) attr: h323-connect-time (59) attr: h323-disconnect-time (64) attr: h323-gw-id (65) attr: h323-setup-time (76) attr: h323-voice-quality (78)

Totally 7 attrs defined.

The following example displays details about the template named "cdr1" only.

Router#

show call accounting-template voice cdr1

CDR template cdr1 is running url: tftp://sanjoe/santa/abc/Templates/cdr1.cdr

The last load was successful.

attr: h323-call-origin (56) attr: h323-call-type (57) attr: h323-gw-id (65) attr: subscriber (79) attr: in-portgrp-id (80) attr: out-portgrp-id (81)

Totally 6 attrs defined.

The following example displays all 64 attributes that can be filtered by a template.

Router#

show call accounting-template voice master

h323-call-origin h323-call-type h323-gw-id h323-setup-time h323-connect-time h323-disconnect-time h323-disconnect-cause

.

.

.

calling-party-category originating-line-info charge-number

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transmission-medium-req redirecting-number backward-call-indicators

Totally 64 attributes are filterable.

The following example displays template activity in the service queue. Initially, no templates are in the dirty state.

Router#

show call accounting-template voice qdump

name url is_dirty no_of_legs

========================================================================= cdr1 tftp://sanjoe/santa/abc 0 cdr2 cdr3 tftp://sanjoe/santa/abc tftp://sanjoe/santa/abc

0

0

After the templates are reloaded during active calls, the display below shows the templates named "cdr1" and

"cdr2" to be in a dirty state.

.

.

.

Templates in freeq cdr1 cdr2 tftp://sanjoe/santa/abc tftp://sanjoe/santa/abc dirty dirty

1

1

The following example displays a summary of all configured accounting templates. The template named

"cdr3" is not in running mode, either because it has been rejected or because it does not exist at the given

URL.

Router#

show call accounting-template voice summary

name url last_load is_running

========================================================================= cdr1 tftp://sanjoe/santa/abc success is running cdr2 cdr3 tftp://sanjoe/santa/abc tftp://sanjoe/santa/abc success fail is running is not running

The table below describes the fields shown in the show call accounting-template voice display.

Table 12: show call accounting-template voice Field Descriptions

Field

name url last_load is_running is_dirty no_of_legs attr

Description

Name of the accounting template.

Location of the accounting template.

Describes if the accounting template was successfully or unsuccessfully loaded from its location.

Describes if the accounting template was activated after it was successfully loaded from its location.

Shows that the accounting template was reloaded during an active call.

Number of call legs.

Vendor-specific attributes (VSAs) defined in an accounting template.

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Related Commands

Command gw-accounting aaa show call accounting-template voice

Description

Configures a new accounting template.

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show call active fax

To display call information for T.37 store-and-forward fax transmissions in progress, use the show call active

fax command in user EXEC or privileged EXEC mode.

show call active fax [brief [id identifier]| compact [duration {less seconds| more seconds}]| id identifier]

Syntax Description brief

id identifier

compact duration

(Optional) Displays a truncated version of fax call information.

(Optional) Displays only the call with the specified

identifier . Range is a hex value from 1 to FFFF.

(Optional) Displays a compact version of the fax call information.

(Optional) Displays active calls that are longer or shorter than a specified seconds value. The arguments and keywords are as follows:

less --Displays calls shorter than the seconds value.

more --Displays calls longer than the seconds value.

seconds --Elapsed time, in seconds. Range is from 1 to 2147483647. There is no default value.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

11.3(1)T

12.0(3)XG

12.0(4)XJ

12.0(4)T

12.0(7)XK

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series.

This command was modified. Support for Voice over Frame Relay (VoFR) was added.

This command was implemented for store-and-forward fax on the Cisco

AS5300.

This command was implemented on the Cisco 7200 series.

This command was implemented on the Cisco MC3810.

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Release

12.1(2)T

12.1(3)T

12.1(5)XM

12.1(5)XM2

12.2(2)XB1

12.2(8)T

12.2(11)T

12.3(14)T

12.4(2)T

12.4(15)T

12.4(16)

12.4(22)T

Modification

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was modified. This command was implemented for modem pass-through over VoIP on the Cisco AS5300.

This command was implemented on the Cisco AS5800.

The command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support was not included for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco

AS5850.

Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and

Cisco AS5850.

This command was modified. T.38 fax relay call statistics were made available to Call Detail Records (CDRs) through vendor-specific attributes (VSAs) and added to the call log.

This command was modified. The LocalHostname display field was added to the VoIP call leg record.

This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

This command was modified. Command output was updated to show IPv6 information.

Usage Guidelines

Use this command to display the contents of the active call table. This command displays information about call times, dial peers, connections, quality of service, and other status and statistical information for T.37

store-and-forward fax calls currently connected through the router. This command works with both on-ramp and off-ramp store-and-forward fax functions.

To display information about fax relay calls in progress, use the show call active voice command.

Examples

The following is sample output from the show call active fax command:

Router#

show call active fax

GENERIC:

SetupTime=22021 ms

Index=1

PeerAddress=peer one

PeerSubAddress=

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PeerId=0

PeerIfIndex=0

LogicalIfIndex=0

ConnectTime=24284

CallState=4

CallOrigin=2

ChargedUnits=0

InfoType=10

TransmitPackets=0

TransmitBytes=0

ReceivePackets=0

ReceiveBytes=41190

MMOIP:

ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]

CallID=1

RemoteIPAddress=10.0.0.0

SessionProtocol=SMTP

SessionTarget=

MessageId=

AccountId=

ImgEncodingType=MH

ImgResolution=fine

AcceptedMimeTypes=2

DiscardedMimeTypes=1

Notification=None

GENERIC:

SetupTime=23193 ms

Index=1

PeerAddress=527....

PeerSubAddress=

PeerId=3469

PeerIfIndex=157

LogicalIfIndex=30

ConnectTime=24284

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=10

TransmitPackets=5

TransmitBytes=6513

ReceivePackets=0

ReceiveBytes=0

TELE:

ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]

CallID=2

Port=3/0/0 (2)

BearerChannel=3/0/0.1

TxDuration=24010 ms

FaxTxDuration=10910 ms

FaxRate=14400

NoiseLevel=-1

ACOMLevel=-1

OutSignalLevel=0

InSignalLevel=0

InfoActivity=0

ERLLevel=-1

SessionTarget=

ImgPages=0

The table below provides an alphabetical listing of the fields displayed in the output of the show call active

faxcommand and a description of each field.

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BearerChannel

Buffer Drain Events

Buffer Fill Events

CallDuration

CallOrigin

CallState

ChargedUnits

CodecBytes

CoderTypeRate

ConnectionId

ConnectTime

Consecutive-packets-lost Events

Corrected packet-loss Events

Dial-Peer

EchoCancellerMaxReflector=64

Table 13: show call active fax Field Descriptions

Field

ACOM Level

Description

Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler, which is the sum of the Echo Return Loss, Echo Return Loss

Enhancement, and nonlinear processing loss for the call.

Identification of the bearer channel carrying the call.

Total number of jitter buffer drain events.

Total number of jitter buffer fill events.

Length of the call, in hours, minutes, and seconds, hh:mm:ss.

Call origin: answer or originate.

Current state of the call.

Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.

Payload size, in bytes, for the codec used.

Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.

Global call identifier for this gateway call.

Time, in milliseconds, at which the call was connected.

Total number of consecutive (two or more) packet-loss events.

Total number of packet-loss events that were corrected using the RFC 2198 method.

Tag of the dial peer sending this call.

The location of the largest reflector, in milliseconds

(ms). The reflector size does not exceed the configured echo path capacity. For example, if 32 ms is configured, the reflector does not report beyond 32 ms.

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Field

ERLLevel

FaxTxDuration

GapFillWithInterpolation

GapFillWithPrediction

GapFillWithRedundancy

GapFillWithSilence

GENERIC

H323 call-legs

HiWaterPlayoutDelay

Index

InfoActivity

InfoType

InSignalLevel

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Description

Current echo return loss (ERL) level for this call.

Duration of fax transmission from this peer to the voice gateway for this call. You can derive the Fax

Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call.

Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1

compression algorithms.

Duration of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.

Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.

Generic or common parameters, that is, parameters that are common for VoIP and telephony call legs.

Total H.323 call legs for which call records are available.

High-water-mark Voice Playout FIFO Delay during this call, in ms.

Dial peer identification number.

Active information transfer activity state for this call.

Information type for this call; for example, voice or fax.

Active input signal level from the telephony interface used by this call.

show aal2 profile through show call filter match-list show call active fax

Field

Last Buffer Drain/Fill Event

LocalHostname

LogicalIfIndex

LoWaterPlayoutDelay

LowerIFName

Media

Modem passthrough signaling method in use

NoiseLevel

OnTimeRvPlayout

OutSignalLevel

PeerAddress

PeerId

Description

Elapsed time since the last jitter buffer drain or fill event, in seconds.

Local hostnames used for locally generated gateway

URLs.

Index number of the logical interface for this call.

Low-water-mark Voice Playout FIFO Delay during this call, in ms.

Physical lower interface information. Appears only if the medium is ATM, Frame Relay (FR), or

High-Level Data Link Control (HDLC).

Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is

TELE. If the call is carried over the voice network side, the entry is either ATM, FR, or HDLC.

Indicates that this is a modem pass-through call and that named signaling events (NSEs)--a

Cisco-proprietary version of named telephone events in RFC 2833--are used for signaling codec upspeed.

The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. This means that you might move to a faster codec when you have both voice and data calls and then slow down when there is only voice traffic.

Active noise level for this call.

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Active output signal level to the telephony interface used by this call.

Destination pattern or number associated with this peer.

ID value of the peer table entry to which this call was made.

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Field

PeerIfIndex

PeerSubAddress

Percent Packet Loss

Port

ReceiveBytes

ReceiveDelay

ReceivePackets

ReleaseSource

RemoteIPAddress

RemoteUDPPort

RoundTripDelay

SelectedQoS

SessionProtocol

SessionTarget

SetupTime

SignalingType

SIP call-legs

Telephony call-legs

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Description

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

Subaddress when this call is connected.

Total percent packet loss.

Identification of the time-division multiplexing

(TDM) voice port carrying the call.

Number of bytes received by the peer during this call.

Average Playout FIFO Delay plus the Decoder Delay during this voice call, in ms.

Number of packets received by this peer during this call.

Number value of the release source.

Remote system IP address for the VoIP call.

Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.

Voice packet round-trip delay between the local and remote systems on the IP backbone for this call.

Selected Resourse Reservation Protocol (RSVP) quality of service (QoS) for this call.

Session protocol used for an Internet call between the local and remote routers through the IP backbone.

Session target of the peer used for this call.

Value of the system UpTime, in milliseconds, when the call associated with this entry was started.

Signaling type for this call; for example, channel-associated signaling (CAS) or common channel signaling (CCS).

Total Session Initiation Protocol (SIP) call legs for which call records are available.

Total telephony call legs for which call records are available.

show aal2 profile through show call filter match-list show call active fax

Field

Time between Buffer Drain/Fills

TransmitBytes

TransmitPackets

TxDuration

VAD

VoiceTxDuration

Description

Minimum and maximum durations between jitter buffer drain or fill events, in seconds.

Number of bytes sent by this peer during this call.

Number of packets sent by this peer during this call.

The length of the call. Appears only if the medium is

TELE.

Whether voice activation detection (VAD) was enabled for this call.

Duration of voice transmission from this peer to the voice gateway for this call, in ms. Derive the Voice

Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.

The following is sample output from the show call active fax brief command:

Router#

show call active fax brief

<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state> \ tx:<packets>/<bytes> rx:<packets>/<bytes> <state>

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late> delay:<last>/<min>/<max>ms <codec>

FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n> sig:<on/off> <codec> (payload size)

Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm

1 : 22021hs.1 +2263 pid:0 Answer wook song active tx:0/0 rx:0/41190

IP 0.0.0.0 AcceptedMime:2 DiscardedMime:1

1 : 23193hs.1 +1091 pid:3469 Originate 527.... active tx:10/13838 rx:0/0

Tele : tx:31200/10910/20290ms noise:-1 acom:-1 i/0:0/0 dBm

The following is sample output from the show call active fax command displaying T.38 fax relay statistics:

Router#

show call active fax

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 0

MGCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

GENERIC:

SetupTime=1874690 ms

Index=1

PeerAddress=5551234

PeerSubAddress=

PeerId=3

PeerIfIndex=244

LogicalIfIndex=118

ConnectTime=187875

CallDuration=00:00:44 sec

CallState=4

CallOrigin=2

ChargedUnits=0

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InfoType=fax

TransmitPackets=309

TransmitBytes=5661

ReceivePackets=1124

ReceiveBytes=49189

TELE:

ConnectionId=[0x6B241E98 0xA78111D8 0x8002000A 0xF4107CA0]

IncomingConnectionId=[0x6B241E98 0xA78111D8 0x8002000A 0xF4107CA0]

CallID=1

Port=3/0/0 (1)

BearerChannel=3/0/0.1

TxDuration=2840 ms

VoiceTxDuration=0 ms

FaxTxDuration=0 ms

FaxRate=disable bps

FaxRelayMaxJitBufDepth 346

FaxRelayJitterBufOverflow 0

Initial HS Modulation is V.17/long/14400

Recent HS modulation is V.17/short/14400

Number of pages 1

Direction of transmission is Transmit

Num of Packets TX'ed/RX'ed 932/52

Packet loss conceal is 0

Encapsulation protocol is T.38 (UDPTL)

ECM is DISABLED

NoiseLevel=0

ACOMLevel=0

OutSignalLevel=0

InSignalLevel=0

InfoActivity=0

ERLLevel=0

SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=5551234

OriginalCallingOctet=0x80

OriginalCalledNumber=5555678

OriginalCalledOctet=0x80

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=5551234

TranslatedCallingOctet=0x80

TranslatedCalledNumber=5555678

TranslatedCalledOctet=0x80

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=5555678

GwReceivedCalledOctet3=0x80

GwReceivedCallingNumber=5551234

GwReceivedCallingOctet3=0x80

GwReceivedCallingOctet3a=0x0

DSPIdentifier=1/0:0

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 0

MGCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

The table below provides an alphabetical listing of the fields displayed in the output of the show call active

faxcommand for T.38 fax relay statistics and a description of each field.

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Table 14: show call active fax Field Descriptions for Significant T.38 Fax Relay Statistics

Field

ACOMLevel

BearerChannel

ERLLevel

FaxRate

FaxRelayJitterBufOverflow

FaxRelayMaxJitBufDepth

FaxTxDuration

Description

Current ACOM level estimate in 0.1 dB increments.

The term ACOM is used in G.165, General

Characteristics of International Telephone

Connections and International Telephone Circuits:

Echo Cancellers . ACOM is the combined loss achieved by the echo canceller, which is the sum of the ERL, ERL enhancement, and nonlinear processing loss for the call.

Identification of the bearer channel carrying the call.

Current ERL level estimate in 0.1 dB increments.

Fax transmission rate from this peer to the specified dial peer, in bits per second (bps).

Fax relay jitter buffer overflow, in ms.

Fax relay maximum jitter buffer depth, in ms.

Duration of fax transmission from this peer to the voice gateway for this call, in ms.

Call information received at the gateway.

GwReceivedCalledNumber,

GwReceivedCalledOctet3

H323 call-legs

Initial HS Modulation

LogicalIfIndex

MGCP call-legs

Multicast call-legs

OriginalCallingNumber, OriginalCalling Octet,

OriginalCalledNumber, OriginalCalledOctet,

OriginalRedirectCalledNumber,

OriginalRedirectCalledOctet

PeerIfIndex

Type of call: H.323.

Initial high speed modulation used.

Index number of the logical interface for this call.

Type of call: Media Gateway Control Protocol

(MGCP).

Type of call: Multicast.

Original call information regarding calling, called, and redirect numbers, and octet-3s. Octet-3s are information elements (IEs) of Q.931 that include type of number, numbering plan indicator, presentation indicator, and redirect reason information.

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

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Field

Port

Recent HS Modulation

SIP call-legs

Telephony call-legs

Total call-legs

Description

Identification of the TDM voice port carrying the call.

Most recent high-speed modulation used.

Type of call: SIP.

Type of call: Telephony.

Total calls.

TranslatedCallingNumber, TranslatedCallingOctet,

TranslatedCalledNumber, TranslatedCalledOctet,

TranslatedRedirectCalledNumber,

TranslatedRedirectCalledOctet

Translated call information.

TxDuration Duration of transmit path open from this peer to the voice gateway for this call, in ms.

VoiceTxDuration Duration of voice transmission from this peer to the voice gateway for this call, in ms.

Related Commands

Command show call active voice show call history show call-router routes show call-router status show voice port

Description

Displays call information for voice calls that are in progress.

Displays the call history table.

Displays the dynamic routes in the cache of the BE.

Displays the Annex G BE status.

Displays configuration information about a specific voice port.

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show call active media

To display call information for media calls in progress, use the show call active media command in user

EXEC or privileged EXEC mode.

show call active media [[brief] [id identifier]| compact [duration {less seconds| more seconds}]]

Syntax Description brief id

identifier

compact duration

less seconds

more seconds

(Optional) Displays a truncated version of call information.

(Optional) Displays only the call with the specified

identifier. The range is a hexadecimal value from 1 to FFFF.

(Optional) Displays a compact version of call information.

(Optional) Displays the call history for the specified time duration.

(Optional) Displays the call history for shorter duration calls, in seconds. The range is from 1 to

2147483647.

(Optional) Displays the call history for longer duration calls, in seconds. The range is from 1 to 2147483647.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

12.4(15)T

12.4(18)M

Modification

This command was introduced.

This command was modified. The lesskeyword, morekeyword, and

seconds argument were added.

Usage Guidelines

Use this command to display the contents of the active call table. This command displays information about call times, dial peers, connections, quality of service, and other status and statistical information for media calls currently connected through the router.

When a media call is no longer active, its record is stored. You can display the record by using the show call

history mediacommand.

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Examples

The following is sample output from the show call active mediacommand:

Router#

show call active media

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Media call-legs: 2

Total call-legs: 2

GENERIC:

SetupTime=408040 ms

Index=1

PeerAddress=sip:[email protected]:5060

PeerSubAddress=

PeerId=2235

PeerIfIndex=185

LogicalIfIndex=0

ConnectTime=408130 ms

CallDuration=00:00:01 sec

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=0

TransmitBytes=0

ReceivePackets=57

ReceiveBytes=9120

VOIP-MEDIA:

ConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

IncomingConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

CallID=18

RemoteIPAddress=10.5.18.224

RemoteUDPPort=10000

RemoteSignallingIPAddress=10.5.18.224

RemoteSignallingPort=5060

RemoteMediaIPAddress=10.5.18.224

RemoteMediaPort=10000

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=rtp-nte

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2

[email protected]

SessionTarget=10.5.18.224

OnTimeRvPlayout=0

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=0 ms

LoWaterPlayoutDelay=0 ms

TxPakNumber=0

TxSignalPak=0

TxComfortNoisePak=0

TxDuration=0

TxVoiceDuration=0

RxPakNumber=0

RxSignalPak=0

RxComfortNoisePak=0

RxDuration=0

RxVoiceDuration=0

RxOutOfSeq=0

RxLatePak=0

RxEarlyPak=0

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RxBadProtocol=0

PlayDelayCurrent=0

PlayDelayMin=0

PlayDelayMax=0

PlayDelayClockOffset=0

PlayDelayJitter=0

PlayErrPredictive=0

PlayErrInterpolative=0

PlayErrSilence=0

PlayErrBufferOverFlow=0

PlayErrRetroactive=0

PlayErrTalkspurt=0

OutSignalLevel=0

InSignalLevel=0

LevelTxPowerMean=0

LevelRxPowerMean=0

LevelBgNoise=0

ERLLevel=0

ACOMLevel=0

ErrRxDrop=0

ErrTxDrop=0

ErrTxControl=0

ErrRxControl=0

Source tg label=test5

ReceiveDelay=0 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

TextRelay = off

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160

Media Setting=flow-through

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=4085254655

TranslatedCallingOctet=0x21

TranslatedCalledNumber=

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwOutpulsedCallingNumber=4085254655

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurCallTimestamp=

LongDurcallDuration=

Username=

GENERIC:

SetupTime=408050 ms

Index=1

PeerAddress=sip:[email protected]:5060

PeerSubAddress=

PeerId=2234

PeerIfIndex=184

LogicalIfIndex=0

ConnectTime=408160 ms

CallDuration=00:00:03 sec

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=188

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TransmitBytes=30080

ReceivePackets=0

ReceiveBytes=0

VOIP-MEDIA:

ConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

IncomingConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

CallID=19

RemoteIPAddress=10.5.18.224

RemoteUDPPort=10002

RemoteSignallingIPAddress=10.5.18.224

RemoteSignallingPort=5060

RemoteMediaIPAddress=10.5.18.224

RemoteMediaPort=10002

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=rtp-nte

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2

[email protected]

SessionTarget=10.5.18.224

OnTimeRvPlayout=1000

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=1495 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=100 ms

LoWaterPlayoutDelay=95 ms

TxPakNumber=0

TxSignalPak=0

TxComfortNoisePak=0

TxDuration=0

TxVoiceDuration=0

RxPakNumber=0

RxSignalPak=0

RxComfortNoisePak=0

RxDuration=0

RxVoiceDuration=0

RxOutOfSeq=0

RxLatePak=0

RxEarlyPak=0

RxBadProtocol=0

PlayDelayCurrent=0

PlayDelayMin=0

PlayDelayMax=0

PlayDelayClockOffset=0

PlayDelayJitter=0

PlayErrPredictive=0

PlayErrInterpolative=0

PlayErrSilence=0

PlayErrBufferOverFlow=0

PlayErrRetroactive=0

PlayErrTalkspurt=0

OutSignalLevel=0

InSignalLevel=0

LevelTxPowerMean=0

LevelRxPowerMean=0

LevelBgNoise=0

ERLLevel=0

ACOMLevel=0

ErrRxDrop=0

ErrTxDrop=0

ErrTxControl=0

ErrRxControl=0

Source tg label=test5

ReceiveDelay=100 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

TextRelay = off

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VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160

Media Setting=flow-through

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=4085254655

TranslatedCallingOctet=0x21

TranslatedCalledNumber=

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwOutpulsedCallingNumber=4085254655

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurCallTimestamp=

LongDurcallDuration=

Username=

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Media call-legs: 2

Total call-legs: 2

The table below describes the significant fields shown in the display.

Table 15: show call active media Field Descriptions

Field

Telephony call-legs

SIP call-legs

H323 call-legs

Media

GENERIC

SetupTime

Description

Total telephony call legs for which call records are available.

Total session initiation protocol (SIP) call legs for which call records are available.

Total H.323 call legs for which call records are available.

Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is

TELE. If the call is carried over the voice network side, the entry is either ATM, FR (for Frame Relay), or HDLC (for High-Level Data Link Control).

Generic or common parameters, that is, parameters that are common for VoIP and telephony call legs.

Value of the system UpTime, in milliseconds, when the call associated with this entry was started.

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Field

Index

PeerAddress

PeerId

PeerIfIndex

LogicalIfIndex

ConnectTime

CallDuration

CallOrigin

CallState

ChargedUnits

InfoType

TransmitBytes

TransmitPackets

ReceivePackets

ReceiveBytes

ReceiveDelay

ConnectionId

RemoteIPAddress

RemoteUDPPort

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Description

Dial peer identification number.

Destination pattern or number associated with this peer.

ID value of the peer table entry to which this call was made.

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

Index number of the logical interface for this call.

Time, in milliseconds, at which the call was connected.

Length of the call, in hours, minutes, and seconds, hh:mm:ss.

Call origin: answer or originate.

Current state of the call.

Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.

Information type for this call; for example, voice or fax.

Number of bytes sent by this peer during this call.

Number of packets sent by this peer during this call.

Number of packets received by this peer during this call.

Number of bytes received by the peer during this call.

Average Playout FIFO Delay plus the Decoder Delay during this voice call, in ms.

Global call identifier for this gateway call.

Remote system IP address for the VoIP call.

Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.

show aal2 profile through show call filter match-list

Field

SelectedQoS

SessionTarget

OnTimeRvPlayout

GapFillWithInterpolation

GapFillWithRedundancy

GapFillWithPrediction

GapFillWithSilence

HiWaterPlayoutDelay

LoWaterPlayoutDelay

CodecBytes

CoderTypeRate

InSignalLevel

show call active media

Description

Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.

Session target of the peer used for this call.

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

Duration of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.

Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call.

Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1

compression algorithms.

Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.

High-water-mark Voice Playout FIFO Delay during this call, in ms.

Low-water-mark Voice Playout FIFO Delay during this call, in ms.

Payload size, in bytes, for the codec used.

Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.

Active input signal level from the telephony interface used by this call.

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show call active media

Field

OutSignalLevel

ERLLevel

ACOMLevel

PeerSubAddress

RoundTripDelay

SessionProtocol

TxDuration

VAD

Related Commands

Command show call history media show aal2 profile through show call filter match-list

Description

Active output signal level to the telephony interface used by this call.

Current echo return loss (ERL) level for this call.

Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler, which is the sum of the Echo Return Loss, Echo Return Loss

Enhancement, and nonlinear processing loss for the call.

Subaddress when this call is connected.

Voice packet round-trip delay between the local and remote systems on the IP backbone for this call.

Session protocol used for an Internet call between the local and remote routers through the IP backbone.

The length of the call. Appears only if the medium is

TELE.

Whether voice activation detection (VAD) was enabled for this call.

Description

Displays the call history table.

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show call active total-calls

To display the total number of calls in progress, use the show call active total-calls command in user EXEC or privileged EXEC mode.

show call active total-calls

Syntax Description

This command has no arguments or keywords.

Command Modes

User EXEC (>)

Privileged EXEC (#)

Command History

Release

15.3(3)M

Modification

This command was introduced.

Examples

The following example shows how to view the total number of active calls:

Device#

show call active total-calls

Total Number of Active Calls : 110

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show call active video

To display call information for Signaling Connection Control Protocol (SCCP), Session Initiation Protocol

(SIP), and H.323 video calls in progress, use the show call active video command in user EXEC or privileged

EXEC mode.

show call active video [[brief] [id call-identifier] | compact [duration {less| more} seconds] | echo-canceller

call-id | session-id WORD | stats]

Syntax Description brief

id call-identifier

compact duration less more

seconds

echo-canceller call-id

session-id WORD

stats

(Optional) Displays a truncated version of active video call information.

(Optional) Displays only the video calls with the specified identifier. The range is from 1 to FFFF.

(Optional) Displays a compact version of active video call information.

(Optional) Displays call history for the specified time duration.

Displays call history for shorter duration calls.

Displays call history for longer duration calls.

Time, in seconds. The range is from 1 to 2147483647.

(Optional) Displays information about the state of the extended echo canceller (EC). The range is from 0 to

FFFFFFFF.

(Optional) Displays session identifier details for a specific session-id. WORD specifies a wildcard pattern which is matched against the localUUID,

RemoteUUID or complete Session-Id header string.

A valid wildcard pattern can comprise one or more combination of any of these characters *, [0-9], [a-f],

[A-F].

(Optional) Displays information about DSP statistics and video quality metrics.

Command Modes

User EXEC (>) Privileged EXEC (#)

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Command History

Cisco IOS Release

12.4(4)XC

12.4(9)T

12.4(11)T

12.4(16); 12.4(15)T

15.1(4)M

Cisco IOS 15.6(2)T

Cisco IOS XE Denali

16.3.1

Cisco Product

Cisco Unified CME

4.0

Modification

This command was introduced.

Cisco Unified CME

4.0

This command was integrated into Cisco IOS Release

12.4(9)T.

--

--

This command was modified. Support was added for SIP and H.323 calls.

This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

Cisco Unified CME

8.6

This command was modified. The stats keyword was added.

-This command was modified.

• The session-id keyword was added.

show call active video brief command output was enhanced to show VRF details and session-id details.

show call active video compact command output was enhanced to show VRF details.

Usage Guidelines

Use this command to display the contents of the active video call table.

Before you can query the echo state, you need to know the hexadecimal ID. Use the show call active video

brief command to find the hexadecimal ID.

Examples

The following is sample output from the show call active video briefcommand:

Router #

show call active video brief

<ID>: <CallID> <start>ms.<index> (<start>) +<connect> pid:<peer_id> <dir> <addr> <state> dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> dscp:<packets violation> media:<packets violation> audio tos:<audio tos value> video tos:<video tos value>

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late> delay:<last>/<min>/<max>ms <codec> <textrelay> <transcoded media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time> long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>

LostPacketRate:<%> OutOfOrderRate:<%>

LocalUUID: <%> RemoteUUID: <%>

VRF:<%>

MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected> last <buf event time>s dur:<Min>/<Max>s

FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

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<codec> (payload size)

Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm

MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> speeds(bps): local <rx>/<tx> remote <rx>/<tx>

Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf> bw: <req>/<act> codec: <audio>/<video> tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes> rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0

SIP call-legs: 1

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

11EC : 2 798700ms.1 (*12:49:45.019 IST Tue Jan 12 2016) +2010 pid:441 Answer sipp connected dur 00:00:50 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0

IP 10.64.86.70:6005 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: Off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a

LostPacketRate:0.00 OutOfOrderRate:0.00

LocalUUID : ab30317f1a784dc48ff824d0d3715d86

RemoteUUID : 47755a9de7794ba387653f2099600ef2

VRF: VRF1

Telephony call-legs: 0

SIP call-legs: 1

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

The following is sample output from the show call active video command:

Router#

show call active video

Telephony call-legs: 4

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 2

Multicast call-legs: 0

Total call-legs: 6

GENERIC:

SetupTime=169281770 ms

Index=2

PeerAddress=

PeerSubAddress=

PeerId=0

PeerIfIndex=0

LogicalIfIndex=0

ConnectTime=169281770 ms

CallDuration=01:20:44 sec

CallState=2

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=819728

TransmitBytes=571031017

ReceivePackets=796308

ReceiveBytes=566120602

VOIP:

ConnectionId[0x0 0x0 0x0 0x0]

IncomingConnectionId[0x0 0x0 0x0 0x0]

CallID=85

GlobalCallId=[0x0 0x0 0x0 0x0]

CallReferenceId=25666520

CallServiceType=Video Conference

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RTP Loopback Call=FALSE sessionIDLocaluuid=6f0a93a3a79451aebeb6d83f79a3359f sessionIDRemoteuuid=a55b0f45861551b88f57d1fb5bb23f89

RemoteIPAddress=0.0.0.0 RemoteUDPPort=2000

RemoteSignallingIPAddress=0.0.0.0

RemoteSignallingPort=0

RemoteMediaIPAddress=1.4.211.39

RemoteMediaPort=2000

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=inband-voice

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=other

ProtocolCallId= sipv2

SessionTarget= 1.4.211.39

SafEnabled=FALSE

OnTimeRvPlayout=0

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=0ms

LoWaterPlayoutDelay=0 ms

Video Conferee Statistics ConfereeActualFrameRate=0

ConfereeActualBitrate=934600

ConfereeTotalRxPackets=129853

ConfereeTotalRxBytes=125825024

ConfereeTotalTxPackets=129853

ConfereeTotalTxBytes=125825085

ConfereeTotalPacketsDropped=313

ConfereeCurrentPacketsDropped=0

ConfereeTotalPacketsOutOfOrder=296

ConfereeCurrentPacketsOutOfOrder=0

ConfereeMaxJitter=0 ConfereeCurJitter=0

ConfereeMaxDelay=0

ConfereeCurDelay=0

ConfereeMaxOutOfSyncDelay=0

ConfereeCurrentOutOfSyncDelay=0

ConfereeFastVideoUpdateRate=0

ConfereeVideoDuration=1076

Video Quality Scores RxVideoMOSInstant=78/100 (Good)

RxVideoMOSAverage=70/100 (Good)

VIDEO: VideoTransmitCodec=H264

VideoTransmitPictureWidth=640

VideoTransmitPictureHeight=480

VideoTransmitFrameRate=30

VideoTransmitBitrate=934600 bps

VideoTransmitLevel=2

VideoTransmitProfile=Baseline

VideoTransmitPayloadFormat=RFC3984

VideoTransmitPackets=129853

VideoTransmitBytes=125825085

VideoTransmitDuration=1076 seconds

VideoReceiveCodec=H264

VideoReceivePictureWidth=640

VideoReceivePictureHeight=480

VideoReceiveFrameRate=30

VideoReceiveBitrate=934600 bps

VideoReceiveLevel=2

VideoReceiveProfile=Baseline

VideoReceivePayloadFormat=RFC3984

VideoReceivePackets=129853

VideoReceiveBytes=125825024

VideoReceiveDuration=1076 seconds

VideoCap_Codec=H264

VideoCap_Format=CUSTOM

VideoCap_PictureWidth=640

VideoCap_PictureHeight=480

VideoCap_FrameRate=30

VideoCap_Bitrate=960000 bps

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VideoCap_Level=2

VideoCap_Profile=Baseline

VideoCap_PayloadFormat=RFC3984

VideoLostPackets=0

VideoEarlyPackets=0

VideoLatePackets=0

VideoUsedBandwidth=934600

VideoNumberOfChannels=0

PlayoutMode = undefined

PlayoutInitialDelay=0 ms

ReceiveDelay=0 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

TextRelay = off

VAD =disabled

CoderTypeRate=h264

CodecBytes=0

Media Setting=flow-around

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=

TranslatedCallingOctet=0x0

TranslatedCalledNumber=

TranslatedCalledOctet=0x0

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0x0

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurCallTimestamp=

LongDurcallDuration=

Username= MlppServiceDomainNW=0 (none)

MlppServiceDomainID=

PrecedenceLevel=0

(PRECEDENCE_LEVEL_NONE)

The following shows sample output from the show call active video statscommand:

Router#

show call active video stats

<ID>: <CallID> <start>ms.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr> dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 1

Multicast call-legs: 0

Total call-legs: 1

0 : 5 *10:54:50.661 PDT Tue Jan 11 2011.2 +0 pid:0 Originate connecting dur 00:17:27 tx:126342/122451295 rx:126640/122453063

Video Conferee Statistics

ConfereeActualFrameRate=0 ConfereeActualBitrate=934300

ConfereeTotalRxPackets=126166 ConfereeTotalRxBytes=122282402

ConfereeTotalTxPackets=126166 ConfereeTotalTxBytes=122282463

ConfereeTotalPacketsDropped=295 ConfereeCurrentPacketsDropped=0

ConfereeTotalPacketsOutOfOrder=278 ConfereeCurrentPacketsOutOfOrder=0

ConfereeMaxJitter=0 ConfereeCurJitter=0

ConfereeMaxDelay=0 ConfereeCurDelay=0

ConfereeMaxOutOfSyncDelay=0 ConfereeCurrentOutOfSyncDelay=0

ConfereeFastVideoUpdateRate=0 ConfereeVideoDuration=1046

Video Quality Scores

RxVideoMOSInstant=78/100 (Good)

(Compression Degradation: 86%, Network Degradation: 13%, Transcoding Degradation: 0%)

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RxVideoMOSAverage=70/100 (Good)

(Compression Degradation: 93%, Network Degradation: 6%, Transcoding Degradation: 0%)

The following is sample output from the show call active video command using the compact keyword:

Router#

show call active video compact

<callID> A/O FAX T<sec> Codec type

Total call-legs: 2

10193925 ANS T22

10193927 ORG T22

Peer

H.264 VOIP-VIDEO P2005

H.264 VOIP-VIDEO P3001

Address IP R<ip>:<udp> VRF

10.0.0.1:18070

11.0.0.1:27008

VRF1

VRF2

The table below describes the significant fields shown in the display, in alphabetical order.

Table 16: show call active video Field Descriptions

Field

CallDuration

CallState

Call agent controlled call-legs

ChargedUnits

CodecBytes

CoderTypeRate

ConnectionId

ConnectTime

EchoCancellerMaxReflector

ERLLevel

FaxTxDuration

Description

Length of the call, in hours, minutes, and seconds, hh:mm:ss.

Current state of the call.

Displays call legs for devices that are not telephony endpoints; for example, transcoding and conferencing

Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of a second.

Payload size, in bytes, for the codec used.

Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.

Global call identifier for this gateway call.

Time, in milliseconds (ms), during which the call was connected.

Size of the largest reflector, in ms. The reflector size cannot exceed the configured echo path capacity. For example, if 32 ms is configured, the reflector does not report capacity beyond 32 ms.

Current echo return loss (ERL) level for this call.

Duration, in ms, of fax transmission from this peer to the voice gateway for this call. You can derive the

Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

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show call active video

Field

GapFillWithInterpolation

GapFillWithRedundancy

GapFillWithPrediction

GapFillWithSilence

GENERIC

H320CallType

H323 call-legs

HiWaterPlayoutDelay

Index

InfoActivity

InfoType

InSignalLevel

Last Buffer Drain/Fill Event

LocalHostname

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Description

Duration, in ms, of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

Duration, in ms, of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.

Duration, in ms, of the voice signal played out with a signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and

G.723.1 compression algorithms.

Duration, in ms, of a voice signal replaced with silence because voice data was lost or not received in time for this call.

Generic or common parameters, that is, parameters that are common for VoIP and telephony call legs.

Total H320 call types available.

Total H.323 call legs for which call records are available.

High-water-mark voice playout first in first out

(FIFO) delay during this call, in ms.

Dial peer identification number.

Active information transfer activity state for this call.

Information type for this call; for example, voice, speech, or fax.

Active input signal level from the telephony interface used by this call.

Elapsed time since the last jitter buffer drain or fill event, in seconds.

Local hostnames used for locally generated gateway

URLs.

show aal2 profile through show call filter match-list

OutSignalLevel

PeerAddress

PeerId

PeerIfIndex

PeerSubAddress

ReceiveBytes

ReceiveDelay

ReceivePackets

RemoteIPAddress

Field

LogicalIfIndex

LoWaterPlayoutDelay

LowerIFName

Media

Multicast call-legs

NoiseLevel

OnTimeRvPlayout

show call active video

Description

Index number of the logical interface for this call.

Low-water-mark voice playout FIFO delay during this call, in ms.

Physical lower interface information. Appears only if the medium is ATM, Frame Relay (FR), or

High-Level Data Link Control (HDLC).

Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is

TELE. If the call is carried over the voice network side, the entry is either ATM, FR, or HDLC.

Total multicast call legs for which call records are available.

Active noise level for this call.

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Active output signal level to the telephony interface used by this call.

Destination pattern or number associated with this peer.

ID value of the peer table entry to which this call was made.

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

Subaddress when this call is connected.

Number of bytes received by the peer during this call.

Average playout FIFO delay plus the decoder delay during this voice call, in ms.

Number of packets received by this peer during this call.

Remote system IP address for the VoIP call.

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show call active video

Field

RemoteUDPPort

RoundTripDelay

SCCP call-legs

SelectedQoS

SessionIDLocaluuid

SessionIDRemoteuuid

SessionProtocol

SessionTarget

SetupTime

SIP call-legs

Telephony call-legs

Total call-legs

TransmitBytes

TransmitPackets

TxDuration

VAD

VideoCap_Annex

VideoCap_Bitrate

VideoCap_Codec

240

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Description

Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.

Voice packet round-trip delay, in ms, between the local and remote systems on the IP backbone for this call.

Call legs for SCCP telephony endpoints.

Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.

UUID generated from originating user agent.

UUID generated from terminating user agent.

Session protocol used for an Internet call between the local and remote routers through the IP backbone.

Session target of the peer used for this call.

Value of the system UpTime, in milliseconds, when the call associated with this entry was started.

Total SIP call legs for which call records are available.

Total telephony call legs for which call records are available.

Total number of call legs for the call.

Number of bytes sent by this peer during this call.

Number of packets sent by this peer during this call.

The length of the call. Appears only if the medium is

TELE.

Whether voice activation detection (VAD) was enabled for this call.

Extension of the video stream; for example, annex

D1 and E.

Negotiated bitrate of the video stream; for example,

128000 b/s.

Codec for the active video call.

show aal2 profile through show call filter match-list

Field

VideoCap_Format

VideoCap_FrameRate

VideoCap_PictureHeight

VideoCap_PictureWidth

VideoEarlyPackets

VideoLatePackets

VideoLostPackets

VideoNumberOfChannels

Video Quality Score

VideoReceiveBytes

VideoReceiveCodec

VideoReceivePackets

VideoTransmitBytes

VideoTransmitCodec

VideoTransmitPackets

VideoUsedBandwidth

show call active video

Description

Video format for the active video call.

Negotiated frame rate of the video stream; for example, 15 or 30 f/s.

Height of the video resolution.

Width of the video resolution.

Number of early packets for a video call.

Number of late packets in a video call.

Number of lost packets in a video call.

Number of channels used for a video call.

Instantaneous and average Mean Opinion Score

(MOS) for each active call leg. The MOS score is based on the amount of video quality degradation caused by compression distortion and the amount of video quality degradation caused by packet loss. The scale for the MOS score is as follows:

• Excellent--(80--100)

• Good--(60--80)

• Fair--(40--60)

• Poor--(20--40)

• Bad--(0--20)

Number of bytes received in the video call.

Type of video codec used in the receiving stream.

Number of packets received in the video call.

Number of bytes transmitted in the video call.

Type of video codec used in the transmission stream.

Number of packets transmitted in the video call.

Bandwidth, in kbps, used for a video call.

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Field

VoiceTxDuration

Related Commands

Command show call history video show aal2 profile through show call filter match-list

Description

Duration of voice transmission from this peer to the voice gateway for this call, in milliseconds. Derive the Voice Utilization Rate by dividing the

VoiceTxDuration value by the TxDuration value.

Description

Displays call history information for SCCP video calls.

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show call active voice

To display call information for voice calls in progress, use the show call active voice command in user EXEC or privileged EXEC mode.

show call active voice [[brief] [long-dur-call-inactive| media-inactive] [called-number number|

calling-number number] [id call-identifier] | compact [duration {less | more} seconds] | dest-route-string

tag | echo-canceller {hexadecimal-id | port slot-number | summary}| long-dur-call [called-number number

| calling-number number ] | redirect tbct | session-id word | stats]

Syntax in Cisco IOS Release 12.2(33)SXH and Subsequent 12.2SX Releases show call active [brief]

Syntax Description brief long-dur-call-inactive media-inactive

called-number number

calling-number number

id

call-identifier

compact duration

less seconds

more seconds

dest-route-string tag

echo-canceller

(Optional) Displays a truncated version of call information.

(Optional) Displays long duration calls that are detected and notified.

(Optional) Displays information about inactive media that have been detected.

(Optional) Displays a specific called number pattern.

(Optional) Displays a specific calling number pattern.

(Optional) Displays only the call with the specified

call-identifier value. The range is from 1 to FFFF.

(Optional) Displays a compact version of call information.

(Optional) Displays the call history for the specified time duration.

Displays the call history for shorter duration calls, in seconds. The range is from 1 to 2147483647.

Displays the call history for longer duration calls, in seconds. The range is from 1 to 2147483647.

(Optional) Displays only the call with the specified destination route tag value. The range is from 1 to

10000.

(Optional) Displays information about the state of the extended echo canceller (EC).

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hexadecimal-id

port slot-number

summary long-dur-call redirect tbct

session-id WORD

stats

The hexadecimal ID of an active voice call. The range is from 0x0 to 0xFFFFFFFF.

Displays EC details for a specified active voice port.

The range varies depending on the voice ports available on the router.

Displays an EC summary for all active voice calls.

(Optional) Displays long duration calls that are detected and notified.

(Optional) Displays information about active calls that are being redirected using Release-to-Pivot

(RTPvt) or Two B-Channel Transfer (TBCT).

Displays information about TBCT calls.

(Optional) Displays session identifier details for a specific session-id. WORD specifies a wildcard pattern which is matched against the localUUID,

RemoteUUID or complete Session-Id header string.

A valid wildcard pattern can comprise one or more combination of any of these characters *, [0-9], [a-f],

[A-F].

(Optional) Displays information about digital signal processing (DSP) voice quality metrics.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

11.3(1)T

12.0(3)XG

12.0(4)XJ

12.0(4)T

12.0(7)XK

12.1(3)T

Modification

This command was introduced.

This command was modified. Support for Voice over Frame Relay (VoFR) was added.

This command was implemented for store-and-forward fax on the Cisco

AS5300.

This command was implemented on the Cisco 7200 series.

This command was implemented on the Cisco MC3810.

This command was implemented for modem pass-through over VoIP on the

Cisco AS5300.

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Release

12.1(5)XM

12.1(5)XM2

12.2(2)XB1

12.2(8)T

12.2(11)T

12.2(13)T

12.3(1)

12.3(4)T

Modification

This command was implemented on the Cisco AS5800.

The command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support was not included for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco

AS5850.

Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and

Cisco AS5850.

This command was modified. The echo-cancellerkeyword was added. The command output was modified with an extra reflector location when the extended EC is present; the largest reflector location is shown.

This command was modified. The redirect keyword was added.

This command was modified. The called-number, calling-number, and

media-inactivekeywords were added.

12.3(14)T

12.4(2)T

This command was modified. New output relating to Skinny Client Control

Protocol (SCCP), SCCP Telephony Control Application (STCAPP), and modem pass-through traffic was added.

This command was modified. The LocalHostname display field was added to the VoIP call leg record and command output was enhanced to display modem relay physical layer and error correction protocols.

This command was modified. The long-dur-call keyword was added.

This command was modified. The stats keyword was added.

12.4(4)T

12.4(11)XW

12.4(15)T This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

This command was integrated into Cisco IOS Release 12.2(33)SXH.

12.2(33)SXH

12.4(16)

12.4(22)T

This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

This command was modified. Command output was updated to show IPv6 information.

This command was modified. The dest-route-string keyword was added.

15.3(3)M

Cisco IOS XE Release 3.10S

This command was integrated into Cisco IOS XE Release 3.10S.

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Release Modification

Cisco IOS 15.6(2)T

Cisco IOS XE Denali 16.3.1

This command was modified.

• The session-id keyword was added.

show call active voice brief command output was enhanced to show

VRF details and session-id details.

show call active voice compact command output was enhanced to show VRF details.

Usage Guidelines

Use this command to display the contents of the active voice call table. This command displays information about call times, dial peers, connections, and quality of service, and other status and statistical information for voice calls currently connected through the router.

Before you can query the echo state, you need to know the hexadecimal ID. To find the hexadecimal ID, enter the show call active voice brief command or use the show voice call statuscommand.

When the extended EC is present, the show call active voice command displays the contents of the Ditech

EC_CHAN_CTRL structure. The table below contains names and descriptions of the fields in the

EC_CHAN_CTRL structure. The table also provides a listing of the information types associated with this command.

Use the show call active voice dest-route-string command to display only the active voice calls with call routing configured using specified destination-route-string globally and in dial-peer level.

Table 17: EC_CHAN_CTRL Field Descriptions

Symbol

BYP0

Field

Channel bypass

Description

• 1 = Transparent bypass; EC is disabled.

• 0 = Cancel; EC is enabled.

TAIL3 Max tail

Note

• 0 = 24 milliseconds.

• 1 = 32 milliseconds.

• 2 = 48 milliseconds.

• 3 = 64 milliseconds.

This field should be set just greater than the anticipated worst round-trip tail delay.

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Symbol

REC3

FRZ0

HZ0

TD3

ERL0

Field

Residual echo control h-register hold h-register clear

Modem tone disable

Echo return loss

show call active voice

Description

• 0 = Cancel only; echo is the result of linear processing; no nonlinear processing is applied.

• 1 = Suppress residual; residual echo is zeroed; simple nonlinear processing is applied (you might experience "dead air" when talking).

• 2 = Reserved.

• 3 = Generate comfort noise

(default).

1 = Freezes h-register; used for testing.

Sending the channel command with this bit set clears the h-register.

• 0 = Ignore 2100 Hz modem answer tone.

• 1 = G.164 mode (bypass canceller if 2100 Hz tone).

• 2 = R.

• 3 = G.165 mode (bypass canceller for phase reversing tone only).

• 0 = 6 decibel (dB).

• 1 = 3 dB.

• 2 = 0 dB.

• 3 = R. Worst echo return loss

(ERL) situation in which canceller still works.

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Examples

Symbol

HLC1

R0

Field

High level compensation

Reserved

Description

• 0 = No attenuation.

• 1 = 6 dB if clipped. On loud circuits, the received direction can be attenuated 6 dB if clipping is observed.

Must be set to 0 to ensure compatibility with future releases.

Use the show call active voice redirect tbctcommand to monitor any active calls that implement RTPvt or

TBCT.

When a call is no longer active, its record is stored. You can display the record by using the show call history

voicecommand.

The following is sample output from the show call active voice command for modem relay traffic:

Router#

show call active voice

Modem Relay Local Rx Speed=0 bps

Modem Relay Local Tx Speed=0 bps

Modem Relay Remote Rx Speed=0 bps

Modem Relay Remote Tx Speed=0 bps

Modem Relay Phy Layer Protocol=v34

Modem Relay Ec Layer Protocol=v14

SPRTInfoFramesReceived=0

SPRTInfoTFramesSent=0

SPRTInfoTFramesResent=0

SPRTXidFramesReceived=0

SPRTXidFramesSent=0

SPRTTotalInfoBytesReceived=0

SPRTTotalInfoBytesSent=0

SPRTPacketDrops=0

The table below describes the significant fields shown in the display.

Table 18: show show call active voice Field Descriptions

Field

Modem Relay Local Rx Speed

Modem Relay Local Tx Speed

Modem Relay Remote Rx Speed

Modem Relay Remote Tx Speed

Modem Relay Phy Layer Protocol

Modem Relay Ec Layer Protocol

Description

Download speed, in bits per second, of the local modem relay.

Upload speed of the local modem relay.

Download speed of the remote modem relay.

Upload speed of the remote modem relay.

Physical protocol of the modem relay.

EC layer protocol of the modem relay.

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Field

SPRTInfoFramesReceived

SPRTInfoTFramesSent

SPRTInfoTFramesResent

SPRTXidFramesReceived

SPRTXidFramesSent

SPRTTotalInfoBytesReceived

SPRTTotalInfoBytesSent

SPRTPacketDrops

Description

Total number of simple packet relay transport (SPRT) protocol frames received.

Total number of SPRT frames sent.

Total number of SPRT frames sent again.

Total number of SPRTS ID frames received.

Total number of SPRTS ID frames sent.

Total number of SPRT bytes received.

Total number of SPRT bytes sent.

Total number of SPRT packets dropped.

The following is sample output from the show call active voice command:

Router#

show call active voice

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 1

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

GENERIC:

SetupTime=1072620 ms

Index=1

PeerAddress=9193927582

PeerSubAddress=

PeerId=8

PeerIfIndex=19

LogicalIfIndex=0

ConnectTime=1078940 ms

CallDuration=00:00:51 sec

CallState=4

CallOrigin=2

ChargedUnits=0

InfoType=speech

TransmitPackets=1490

TransmitBytes=0

ReceivePackets=2839

ReceiveBytes=56780

VOIP:

ConnectionId[0xE28B6D1D 0x3D9011D6 0x800400D0 0xBA0D97A1]

IncomingConnectionId[0xE28B6D1D 0x3D9011D6 0x800400D0 0xBA0D97A1]

CallID=1 sessionIDLocaluuid=4fd24d9121935531a7f8d750ad16e19 sessionIDRemoteuuid=db248b6cbdc547bbc6c6fdfb6916eeb

RemoteIPAddress=10.44.44.44 RemoteUDPPort=17096

RemoteSignallingIPAddress=10.44.44.44

RemoteSignallingPort=56434

RemoteMediaIPAddress=10.44.44.44

RemoteMediaPort=17096

RoundTripDelay=6 ms

SelectedQoS=best-effort tx_DtmfRelay=h245-signal

FastConnect=TRUE AnnexE=FALSE

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Separate H245 Connection=FALSE

H245 Tunneling=TRUE SessionProtocol=cisco

ProtocolCallId= sipv2

SessionTarget= 10.44.44.44

OnTimeRvPlayout=54160

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=70 ms

LoWaterPlayoutDelay=60 ms

TxPakNumber=1490

TxSignalPak=0

TxComfortNoisePak=1

TxDuration=54240

TxVoiceDuration=29790

RxPakNumber=2711

RxSignalPak=0

RxDuration=0

TxVoiceDuration=54210

VoiceRxDuration=54160

RxOutOfSeq=0

RxLatePak=0

RxEarlyPak=0

PlayDelayCurrent=60

PlayDelayMin=60

PlayDelayMax=70

PlayDelayClockOffset=212491899

PlayDelayJitter=0 ms

PlayErrPredictive=0

PlayErrInterpolative=0

PlayErrSilence=0

PlayErrBufferOverFlow=10

PlayErrRetroactive=0

PlayErrTalkspurt=0

OutSignalLevel=-57

InSignalLevel=-51

LevelTxPowerMean=0

LevelRxPowerMean=-510

LevelBgNoise=0

ERLLevel=16

ACOMLevel=16

ErrRxDrop=0

ErrTxDrop=0

ErrTxControl=0

ErrRxControl=0

ReceiveDelay=60 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

VAD =enabled

CoderTypeRate=g729r8

CodecBytes=20

Media Setting=flow-through

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=9193927582

OriginalCallingOctet=0x21

OriginalCalledNumber=93615494

OriginalCalledOctet=0xC1

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=9193927582

TranslatedCallingOctet=0x21

TranslatedCalledNumber=93615494

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=93615494

GwReceivedCalledOctet3=0xC1

GwReceivedCallingNumber=9193927582

GwReceivedCallingOctet3=0x21

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GwReceivedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

Username= GENERIC:

SetupTime=1072760 ms

Index=1 PeerAddress=93615494

PeerSubAddress=

PeerId=9

PeerIfIndex=18

LogicalIfIndex=4

ConnectTime=1078940 ms

CallDuration=00:00:53 sec

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=2953

TransmitBytes=82684

ReceivePackets=1490

ReceiveBytes=29781 TELE:

ConnectionId=[0xE28B6D1D 0x3D9011D6 0x800400D00xBA0D97A1]

IncomingConnectionId=[0xE28B6D1D 0x3D9011D6 0x800400D0 0xBA0D97A1]

CallID=2

Port=3/0/0 (1)

BearerChannel=3/0/0.2

TxDuration=59080 ms

VoiceTxDuration=29790 ms

FaxTxDuration=0 ms

CoderTypeRate=g729r8

NoiseLevel=-54

ACOMLevel=16

OutSignalLevel=-57

InSignalLevel=-51

InfoActivity=1

ERLLevel=16

EchoCancellerMaxReflector=8

SessionTarget= ImgPages=0 CallerName=

CallerIDBlocked=False

AlertTimepoint=1073340 ms

OriginalCallingNumber=9193927582

OriginalCallingOctet=0x21

OriginalCalledNumber=93615494

OriginalCalledOctet=0xC1

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=9193927582

TranslatedCallingOctet=0x21

TranslatedCalledNumber=93615494

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=93615494

GwReceivedCalledOctet3=0xC1

GwOutpulsedCalledNumber=93615494

GwOutpulsedCalledOctet3=0xC1

GwReceivedCallingNumber=9193927582

GwReceivedCallingOctet3=0x21

GwReceivedCallingOctet3a=0x81

GwOutpulsedCallingNumber=9193927582

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

DSPIdentifier=3/1:1

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 1

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

show call active voice

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The first table above and the table below describe the significant fields shown in the display, in alphabetical order.

Table 19: show call active voice Field Descriptions

Field

CallDuration

CallState

Call agent controlled call-legs

ChargedUnits

CodecBytes

CoderTypeRate

ConnectionId

ConnectTime

EchoCancellerMaxReflector

ERLLevel

FaxTxDuration

GapFillWithInterpolation

GapFillWithRedundancy

Description

Length of the call, in hours, minutes, and seconds, hh:mm:ss.

Current state of the call.

Displays call legs for devices that are not telephony endpoints; for example, transcoding and conferencing

Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.

Payload size, in bytes, for the codec used.

Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.

Global call identifier for this gateway call.

Time, in ms, during which the call was connected.

Size of the largest reflector, in ms. The reflector size cannot exceed the configured echo path capacity. For example, if 32 ms is configured, the reflector does not report capacity beyond 32 ms.

Current echo return loss (ERL) level for this call.

Duration, in ms, of fax transmission from this peer to the voice gateway for this call. You can derive the

Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

Duration, in ms, of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

Duration, in ms, of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.

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Field

GapFillWithPrediction

GapFillWithSilence

GENERIC

H320CallType

H323 call-legs

HiWaterPlayoutDelay

Index

InfoActivity

InfoType

InSignalLevel

LogicalIfIndex

LoWaterPlayoutDelay

Media

Multicast call-legs

NoiseLevel

show call active voice

Description

Duration, in ms, of the voice signal played out with a signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and

G.723.1 compression algorithms.

Duration, in ms, of a voice signal replaced with silence because voice data was lost or not received in time for this call.

Generic or common parameters; that is, parameters that are common for VoIP and telephony call legs.

Total H320 call types available.

Total H.323 call legs for which call records are available.

High-water-mark voice playout first in first out

(FIFO) delay during this call, in ms.

Dial peer identification number.

Active information transfer activity state for this call.

Information type for this call; for example, voice, speech, or fax.

Active input signal level from the telephony interface used by this call.

Index number of the logical interface for this call.

Low-water-mark voice playout FIFO delay during this call, in ms.

Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is

TELE. If the call is carried over the voice network side, the entry is either ATM, Frame Relay (FR), or

High-Level Data Link Control (HDLC).

Total multicast call legs for which call records are available.

Active noise level for this call.

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Field

OnTimeRvPlayout

OutSignalLevel

PeerAddress

PeerId

PeerIfIndex

PeerSubAddress

ReceiveBytes

ReceiveDelay

ReceivePackets

RemoteIPAddress

RemoteUDPPort

RoundTripDelay

SCCP call-legs

SelectedQoS

SessionIDLocaluuid

SessionIDRemoteuuid

SessionProtocol

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Description

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Active output signal level to the telephony interface used by this call.

Destination pattern or number associated with this peer.

ID value of the peer table entry to which this call was made.

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

Subaddress when this call is connected.

Number of bytes received by the peer during this call.

Average playout FIFO delay plus the decoder delay during this voice call, in ms.

Number of packets received by this peer during this call.

Remote system IP address for the VoIP call.

Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.

Voice packet round-trip delay, in ms, between the local and remote systems on the IP backbone for this call.

Call legs for SCCP telephony endpoints.

Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.

UUID generated from the originating user agent.

UUID generated from the terminating user agent.

Session protocol used for an Internet call between the local and remote routers through the IP backbone.

show aal2 profile through show call filter match-list show call active voice

Field

SessionTarget

SetupTime

SIP call-legs

Telephony call-legs

Total call-legs

TransmitBytes

TransmitPackets

TxDuration

VAD

VoiceTxDuration

Description

Session target of the peer used for this call.

Value of the system UpTime, in ms, when the call associated with this entry was started.

Total SIP call legs for which call records are available.

Total telephony call legs for which call records are available.

Total number of call legs for the call.

Number of bytes sent by this peer during this call.

Number of packets sent by this peer during this call.

The length of the call. Appears only if the medium is

TELE.

Whether voice activation detection (VAD) was enabled for this call.

Duration of voice transmission from this peer to the voice gateway for this call, in ms. Derive the Voice

Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.

The following is sample output from the show call active voice command for voice traffic over call-agent controlled call legs. Note that call legs for SCCP telephony endpoints, that is, phones controlled by STCAPP, are displayed under the "Call agent controlled call-legs" field ("SCCP call-legs" displays call legs for devices that are not telephony endpoints; for example, transcoding and conferencing).

Router#

show call active voice

Telephony call-legs: 2

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 2

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 4

GENERIC:

SetupTime=1557650 ms

Index=1

PeerAddress=

PeerSubAddress=

PeerId=999100

PeerIfIndex=14

LogicalIfIndex=10

ConnectTime=1562040 ms

CallDuration=00:01:01 sec

CallState=4

CallOrigin=2

ChargedUnits=0

InfoType=speech

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TransmitPackets=3101

TransmitBytes=519564

ReceivePackets=3094

ReceiveBytes=494572

TELE:

ConnectionId=[0x11B1860C 0x22D711D7 0x8014E4D4 0x8FD15327]

IncomingConnectionId=[0x11B1860C 0x22D711D7 0x8014E4D4 0x8FD15327]

CallID=25

Port=3/0/0 (25)

BearerChannel=3/0/0.1

TxDuration=59670 ms

VoiceTxDuration=59670 ms

FaxTxDuration=0 ms

CoderTypeRate=g711ulaw

NoiseLevel=-12

ACOMLevel=22

OutSignalLevel=-12

InSignalLevel=-11

InfoActivity=1

ERLLevel=22

EchoCancellerMaxReflector=2

SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x80

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=

TranslatedCallingOctet=0x0

TranslatedCalledNumber=

TranslatedCalledOctet=0x80

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0x0

DSPIdentifier=1/1:1

GENERIC:

SetupTime=1559430 ms

Index=1

PeerAddress=7702

PeerSubAddress=

PeerId=999100

PeerIfIndex=14

LogicalIfIndex=11

ConnectTime=1562020 ms

CallDuration=00:01:03 sec

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=3151

TransmitBytes=528900

ReceivePackets=3158

ReceiveBytes=503876

TELE:

ConnectionId=[0x0 0x0 0x0 0x0]

IncomingConnectionId=[0x0 0x0 0x0 0x0]

CallID=26

Port=3/0/0 (26)

BearerChannel=3/0/0.2

TxDuration=60815 ms

VoiceTxDuration=60815 ms

FaxTxDuration=0 ms

CoderTypeRate=g711ulaw

NoiseLevel=-12

ACOMLevel=28

OutSignalLevel=-12

InSignalLevel=-11

InfoActivity=1

ERLLevel=28

EchoCancellerMaxReflector=2

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SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

AlertTimepoint=1559430 ms

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=7701

TranslatedCallingOctet=0x0

TranslatedCalledNumber=7702

TranslatedCalledOctet=0x0

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0x0

GwOutpulsedCalledNumber=7702

GwOutpulsedCalledOctet3=0x0

GwOutpulsedCallingNumber=7701

GwOutpulsedCallingOctet3=0x0

GwOutpulsedCallingOctet3a=0x0

DSPIdentifier=1/1:2

GENERIC:

SetupTime=1562040 ms

Index=1

PeerAddress=

PeerSubAddress=

PeerId=0

PeerIfIndex=0

LogicalIfIndex=0

ConnectTime=0 ms

CallDuration=00:00:00 sec

CallState=2

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=3215

TransmitBytes=512996

ReceivePackets=3208

ReceiveBytes=512812

VOIP:

ConnectionId[0x0 0x0 0x0 0x0]

IncomingConnectionId[0x0 0x0 0x0 0x0]

CallID=27

RemoteIPAddress=10.10.0.0

RemoteUDPPort=17718

RemoteSignallingIPAddress=10.10.0.0

RemoteSignallingPort=0

RemoteMediaIPAddress=10.2.6.10

RemoteMediaPort=17718

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=inband-voice

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=other

ProtocolCallId=

SessionTarget=

OnTimeRvPlayout=60640

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=105 ms

LoWaterPlayoutDelay=105 ms

TxPakNumber=3040

TxSignalPak=0

TxComfortNoisePak=0

TxDuration=60815

TxVoiceDuration=60815

show call active voice

Cisco IOS Voice Command Reference - S commands

257

show call active voice

RxPakNumber=3035

RxSignalPak=0

RxDuration=0

TxVoiceDuration=60690

VoiceRxDuration=60640

RxOutOfSeq=0

RxLatePak=0

RxEarlyPak=0

PlayDelayCurrent=105

PlayDelayMin=105

PlayDelayMax=105

PlayDelayClockOffset=-1662143961

PlayDelayJitter=0

PlayErrPredictive=0

PlayErrInterpolative=0

PlayErrSilence=0

PlayErrBufferOverFlow=0

PlayErrRetroactive=0

PlayErrTalkspurt=0

OutSignalLevel=-12

InSignalLevel=-11

LevelTxPowerMean=0

LevelRxPowerMean=-115

LevelBgNoise=0

ERLLevel=28

ACOMLevel=28

ErrRxDrop=0

ErrTxDrop=0

ErrTxControl=0

ErrRxControl=0

PlayoutMode = undefined

PlayoutInitialDelay=0 ms

ReceiveDelay=105 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160

Media Setting=flow-around

Modem passthrough signaling method is nse:

Buffer Fill Events = 0

Buffer Drain Events = 0

Percent Packet Loss = 0

Consecutive-packets-lost Events = 0

Corrected packet-loss Events = 0

Last Buffer Drain/Fill Event = 0sec

Time between Buffer Drain/Fills = Min 0sec Max 0sec

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=

TranslatedCallingOctet=0x0

TranslatedCalledNumber=

TranslatedCalledOctet=0x0

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0x0

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

Username=

GENERIC:

SetupTime=1562040 ms

Index=2

PeerAddress=

PeerSubAddress=

PeerId=0

show aal2 profile through show call filter match-list

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PeerIfIndex=0

LogicalIfIndex=0

ConnectTime=0 ms

CallDuration=00:00:00 sec

CallState=2

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=3380

TransmitBytes=540332

ReceivePackets=3386

ReceiveBytes=540356

VOIP:

ConnectionId[0x0 0x0 0x0 0x0]

IncomingConnectionId[0x0 0x0 0x0 0x0]

CallID=28

RemoteIPAddress=10.0.0.0

RemoteUDPPort=18630

RemoteSignallingIPAddress=10.10.0.0

RemoteSignallingPort=0

RemoteMediaIPAddress=10.2.6.10

RemoteMediaPort=18630

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=inband-voice

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=other

ProtocolCallId=

SessionTarget=

OnTimeRvPlayout=63120

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=105 ms

LoWaterPlayoutDelay=105 ms

TxPakNumber=3158

TxSignalPak=0

TxComfortNoisePak=0

TxDuration=63165

TxVoiceDuration=63165

RxPakNumber=3164

RxSignalPak=0

RxDuration=0

TxVoiceDuration=63165

VoiceRxDuration=63120

RxOutOfSeq=0

RxLatePak=0

RxEarlyPak=0

PlayDelayCurrent=105

PlayDelayMin=105

PlayDelayMax=105

PlayDelayClockOffset=957554296

PlayDelayJitter=0

PlayErrPredictive=0

PlayErrInterpolative=0

PlayErrSilence=0

PlayErrBufferOverFlow=0

PlayErrRetroactive=0

PlayErrTalkspurt=0

OutSignalLevel=-12

InSignalLevel=-11

LevelTxPowerMean=0

LevelRxPowerMean=-114

LevelBgNoise=0

ERLLevel=22

ACOMLevel=22

ErrRxDrop=0

ErrTxDrop=0

ErrTxControl=0

show call active voice

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ErrRxControl=0

PlayoutMode = undefined

PlayoutInitialDelay=0 ms

ReceiveDelay=105 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160

Media Setting=flow-around

Modem passthrough signaling method is nse:

Buffer Fill Events = 0

Buffer Drain Events = 0

Percent Packet Loss = 0

Consecutive-packets-lost Events = 0

Corrected packet-loss Events = 0

Last Buffer Drain/Fill Event = 0sec

Time between Buffer Drain/Fills = Min 0sec Max 0sec

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=

TranslatedCallingOctet=0x0

TranslatedCalledNumber=

TranslatedCalledOctet=0x0

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0x0

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

Username=

Telephony call-legs: 2

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 2

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 4

The tables above describe the significant fields shown in the display, in alphabetical order.

The following is sample output from the show call active voice command to indicate if Service Advertisement

Framework (SAF) is being used:

Router#

show call active voice

Total call-legs: 2

GENERIC:

SetupTime=1971780 ms

Index=1

PeerAddress=6046692010

PeerSubAddress=

PeerId=20003

PeerIfIndex=17

.

.

.

VOIP:

SessionProtocol=sipv2

[email protected]

SessionTarget=1.5.6.10

SafEnabled=TRUE

SafTrunkRouteId=1

SafPluginDialpeerTag=8

The tables above describe the significant fields shown in the display.

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The following is sample output from the show call active voice command for fax-relay traffic:

Router#

show call active voice

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 1

MGCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

GENERIC:

SetupTime=1049400 ms

Index=2

PeerAddress=52930

PeerSubAddress=

PeerId=82

PeerIfIndex=222

LogicalIfIndex=0

ConnectTime=105105

CallDuration=00:00:59

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=10

TransmitPackets=1837

TransmitBytes=29764

ReceivePackets=261

ReceiveBytes=4079

VOIP:

ConnectionId[0xEB630F4B 0x9F5E11D7 0x8008CF18 0xB9C3632]

IncomingConnectionId[0xEB630F4B 0x9F5E11D7 0x8008CF18 0xB9C3632]

RemoteIPAddress=10.7.95.3

RemoteUDPPort=16610

RemoteSignallingIPAddress=10.7.95.3

RemoteSignallingPort=1720

RemoteMediaIPAddress=10.7.95.3

RemoteMediaPort=16610

RoundTripDelay=13 ms

SelectedQoS=best-effort tx_DtmfRelay=inband-voice

FastConnect=TRUE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=TRUE

SessionProtocol=cisco

ProtocolCallId=

SessionTarget=ipv4:10.7.95.3

OnTimeRvPlayout=1000

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=110 ms

LoWaterPlayoutDelay=70 ms

ReceiveDelay=70 ms

LostPackets=0

EarlyPackets=1

LatePackets=0

VAD = enabled

CoderTypeRate=t38

CodecBytes=40

Media Setting=flow-through

AlertTimepoint=104972

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=4085550130

OriginalCallingOctet=0x0

OriginalCalledNumber=52930

OriginalCalledOctet=0xE9

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x7F

TranslatedCallingNumber=4085550130

TranslatedCallingOctet=0x0

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TranslatedCalledNumber=52930

TranslatedCalledOctet=0xE9

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=52930

GwReceivedCalledOctet3=0xE9

GwOutpulsedCalledNumber=52930

GwOutpulsedCalledOctet3=0xE9

GwReceivedCallingNumber=555-0100

GwReceivedCallingOctet3=0x0

GwReceivedCallingOctet3a=0x80

GwOutpulsedCallingNumber=555-0101

GwOutpulsedCallingOctet3=0x0

GwOutpulsedCallingOctet3a=0x80

Username=

FaxRelayMaxJitterBufDepth = 0 ms

FaxRelayJitterBufOverFlow = 0

FaxRelayHSmodulation = 0

FaxRelayNumberOfPages = 0

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 1

MGCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

The tables above describe the significant fields shown in the display.

The following is sample output from the show call active voice brief command:

Router#

show call active voice brief

<ID>: <CallID> <start>ms.<index> (<start>) +<connect> pid:<peer_id> <dir> <addr> <state> dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> dscp:<packets violation> media:<packets violation> audio tos:<audio tos value> video tos:<video tos value>

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late> delay:<last>/<min>/<max>ms <codec> <textrelay> <transcoded media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time> long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>

LostPacketRate:<%> OutOfOrderRate:<%>

LocalUUID:<%> RemoteUUID:<%>

VRF:<%>

MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected> last <buf event time>s dur:<Min>/<Max>s

FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm

MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> speeds(bps): local <rx>/<tx> remote <rx>/<tx>

Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf> bw: <req>/<act> codec: <audio>/<video> tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes> rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0

SIP call-legs: 2

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

1218 : 15 2442930ms.1 (*13:22:20.797 UTC Thu Feb 18 2016) +2010 pid:880 Answer sipp active dur 00:00:02 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0

IP 1.4.186.60:6005 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off

Transcoded: No ICE: Off

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media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a

LostPacketRate:0.00 OutOfOrderRate:0.00

LocalUUID:ab30317f1a784dc48ff824d0d3715d86

RemoteUUID:47755a9de7794ba387653f2099600ef2

VRF: VRF1

1218 : 16 2442940ms.1 (*13:22:20.807 UTC Thu Feb 18 2016) +2000 pid:770 Originate 7777 active dur 00:00:02 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0

IP 9.45.33.11:16384 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off

Transcoded: No ICE: Off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a

LostPacketRate:0.00 OutOfOrderRate:0.00

LocalUUID:47755a9de7794ba387653f2099600ef2

RemoteUUID:ab30317f1a784dc48ff824d0d3715d86

VRF: NA

Telephony call-legs: 0

SIP call-legs: 2

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

The following is an example of the show call active voicecommand using the echo-canceller keyword. The number 9 represents the hexadecimal ID of an active voice call.

Router#

show call active voice echo-canceller 9

ACOM=-65 ERL=45

Echo canceller control words=6C 0

Bypass=OFF Tail=64 Residual ecan=Comfort noise

Freeze=OFF Modem tone disable=Ignore 2100Hz tone

Worst ERL=6 High level compensation=OFF

Max amplitude reflector (in msec)=5

Ecan version = 8180

The following is sample output from the show call active voice echo-cancellercommand for a call with a hexadecimal ID of 10:

Router#

show call active voice echo-canceller 10

ACOM=-15 ERL=7

Echo canceller control words=6C 0

Bypass=OFF Tail=64 Residual ecan=Comfort noise

Freeze=OFF Modem tone disable=Ignore 2100Hz tone

Worst ERL=6 High level compensation=OFF

Max amplitude reflector (in msec)=64

The call ID number (which is 10 in the preceding example) changes with every new active call. When an active call is up, you must enter the show call active voice briefcommand to obtain the call ID number. The call ID must be converted to hexadecimal value if you want to use the show call active voice echo-canceller

xcommand (x = call ID converted to hexadecimal value).

The table below shows call ID examples converted to hexadecimal values (generally incremented by 2):

Table 20: Call IDs Converted to Hex

Decimal

2

4

Hex

2

4

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10

12

Decimal

6

8

A

C

Hex

6

8

Alternatively, you can use the show voice call statuscommand to obtain the call ID. The call ID output is already in hexadecimal values form when you use this command:

Router#

show voice call status

CallID

0x1

CID ccVdb Port

11CE 0x02407B20 1:0.1

DSP/Ch Called # Codec

1/1 1000

Dial-peers g711ulaw 2000/1000

The following is sample output from the show call active voice command using the compact keyword:

Router#

show call active voice compact

<callID> A/O FAX T<sec> Codec

Total call-legs: 2

8565722 ANS

8565723 ORG

T12

T12 type Peer Address IP R<ip>:<udp> g711ulaw VOIP P777412373 10.0.0.1:30804 g711ulaw VOIP P777512373 11.0.0.1:30804

VRF

VRF1

VRF2

The following is sample output from the show call active voice redirectcommand using the tbct keyword:

Router#

show call active voice redirect tbct

TBCT:

Maximum no. of TBCT calls allowed:No limit

Maximum TBCT call duration:No limit

Total number TBCT calls currently being monitored = 1 ctrl name=T1-2/0, tag=13, call-ids=(7, 8), start_time=*00:12:25.985 UTC Mon Mar 1 1993

The table below describes the significant fields shown in the display.

Table 21: show call active voice redirect Field Descriptions

Field

Maximum no. of TBCT calls allowed

Description

Maximum number of calls that can use TBCT as defined by the tbct max calls command.

Maximum TBCT call duration Maximum length allowed for a TBCT call as defined by the tbct max call-duration command.

Total number TBCT calls currently being monitored Total number of active TBCT calls.

ctrl name tag call-ids

Name of the T1 controller where the call originated.

Call tag number that identifies the call.

Numbers that uniquely identify the call legs.

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Field

start_time

Related Commands

Command show call active fax show call history show call-router routes show call-router status show dial-peer voice show num-exp show voice call status show voice port show call active voice

Description

Time, in hours, minutes, and seconds, when the redirected call began.

Description

Displays call information for fax transmissions that are in progress.

Displays the call history table.

Displays the dynamic routes in the cache of the BE.

Displays the Annex G BE status.

Displays configuration information for dial peers.

Displays how the number expansions are configured in VoIP.

Displays the call status for voice ports on the Cisco router or concentrator.

Displays configuration information about a specific voice port.

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show call application app-level

To display application-level statistics for voice applications, use the show call application app-levelcommand in privileged EXEC mode.

show call application {active| history} app-level [app-tag application-name| summary]

Syntax Description active history

app-tag application-name

summary

Displays statistics for active application instances.

Displays statistics for terminated application instances.

Name of a specific voice application. Output displays statistics for that voice application.

Displays a summary for each application.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(8)T

Modification

This command was introduced.

Usage Guidelines

• To display statistics with this command, you must enable statistics collection with the call application

stats command.

• This command displays gauges and counters that are aggregated per application. The values represent all instances of a particular voice application running on the gateway while statistics collection is enabled.

• To reset application-level counters to zero and subtract the counters from the gateway-level statistics in history, use the clear call application stats command. Statistic counters continue accumulating in history until you use the clear call application stats command or the gateway reloads.

Note

Statistics for an application are automatically cleared if the application is deleted with the no call

application voice command or its script is reloaded with the call application voice load command.

Examples

The following is sample output from the show call application app-level command using different keywords:

Router#

show call application active app-level summary

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Application level active Info:

Sessions

App Name session fax_hop_on w/ Stats

0

0

Total

0

0 clid_authen clid_authen_collect clid_authen_npw clid_authen_col_npw clid_col_npw_3 clid_col_npw_npw

Default

0

0

0

0

0

0

0

0

0

0

0

0

0

0 lib_off_app fax_on_vfc_onramp_app asr offramp generic smtp_record

0

0

0

0

1

0

0

0

0

0

1

0 authen authorize

0

0

0

0 ram_record_replay 0 0

Router#

show call application active app-level app-tag generic

Application level active Info:

Application Name: url:

Total sessions:

Sessions w/ stats: generic tftp://10.10.10.113/tftplocal/generic.vxml

1

1

Currently connected incoming PSTN legs:

Currently connected outgoing PSTN legs:

Currently connected incoming VoIP legs:

Currently connected outgoing VoIP legs:

Placecalls in transit:

Handouts in transit:

1

0

0

0

0

0

Pending ASNL subscriptions:

Pending ASNL unsubscriptions:

Prompts playing (non-TTS):

Recordings:

TTS prompts playing:

0

0

0

0

0

For a description of the fields shown in the display above, see Table 38 on page 1363.

Router#

show call application history app-level summary

Application level history Info:

App Name session

Sessions

Stats w/ Stats Total

N 0 0 fax_hop_on clid_authen clid_authen_collect clid_authen_npw clid_authen_col_npw clid_col_npw_3 clid_col_npw_npw

N

N

N

N

N

N

N

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Default lib_off_app fax_on_vfc_onramp_app ram_record_replay authorize authen smtp_record generic offramp asr

N

N

N

N

N

N

N

Y

N

N

0

0

0

0

0

0

0

2

0

0

0

2

0

0

0

0

0

0

0

0

The table below describes the fields shown in the display.

0

4

0

0

0

0

0

0

0

0

0

0

0

0

0

Errors

0

0

0

Last Reset

Time

*Jul 3 15:49:28

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Table 22: show call application history app-level Field Descriptions

Field

App Name

Stats

Sessions w/ stats

Total

Errors

Last Reset Time

Description

Name of the voice application.

Whether statistics is enabled for this application.

Note

If statistics is enabled, this field displays N until there is at least one active instance of the application.

Number of terminated application instances that the gauges represent.

Total number of instances of the application.

Total number of errors for all instances of the application.

Time at which the statistics were last cleared with the

clear call application stats command, or the gateway was restarted.

Router#

show call application history app-level app-tag generic

Application level history Info:

Application name: generic

URL: tftp://10.10.10.113/tftplocal/generic.vxml

Total sessions:

Sessions w/ stats:

Last reset time:

Statistics:

Subscriber Service - Call

2

2

*Jul 3 15:49:28 PST

PSTN VOIP

Incoming Outgoing Incoming Outgoing

Legs setup:

Total legs connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

2

2

0

0

0

0

2

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

Application Internal Service - Handoff

Play

3

0

0

3

3

Record

0

0

0

0

0

Bridged handoffs:

Bridged handoffs returned:

Blind handoffs:

Handoffs failed:

Incoming Outgoing

0

0

0 x

Application Internal Service - Placecall/transfer

Placecall requests: 0

Placecall successes: 0

0

0

0

0

0

0

0

TTS

0

0

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Placecall failures: 0

Application Internal Service - Document Read-Write

Doc requests:

Doc successes:

Read

0

0

Write

0

0

0 Doc failures: 0

Application Internal Service - Downloaded Script

Script parse errors:

Application Internal Service - ASNL

0

ASNL notifications: 0

Subscription Unsubscription

0 0 ASNL requests:

ASNL successes:

ASNL failures:

0

0

0

0

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

Subscriber Interaction - ASR

ASRs not matched:

ASRs matched:

ASRs no input:

Subscriber Interaction - AAA

0

0

1

0

0

0

0

AAA successes:

AAA failures:

Authentication Authorization

0 1

0 0

For a description of the fields shown in this display, see Table 41 on page 1379.

Related Commands

Command call application event-log call application stats call application voice event-log clear call application stats show call application gateway-level show call application session-level

Description

Enables event logging for voice application instances.

Enables statistics collection for voice applications.

Enables event logging for a specific voice application.

Clears application-level statistics in history and subtracts the statistics from the gateway-level statistics.

Displays gateway-level statistics for voice application instances.

Displays event logs and statistics for voice application instances.

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show call application gateway-level

To display gateway-level statistics for voice application instances, use the show call application gateway-level command in privileged EXEC mode.

show call application {active| history} gateway-level

Syntax Description active history

Displays statistics for active application instances.

Displays statistics for terminated application instances.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(8)T

Modification

This command was introduced.

Usage Guidelines

• To display statistics with this command, you must enable statistics collection with the call application

stats command.

• This command displays gauges and counters that are aggregated per gateway. The values represent all instances of all voice applications running on the gateway while statistics collection is enabled.

• To reset application-level counters to zero and subtract the counters from the gateway-level statistics in history, use the clear call application stats command. Statistic counters continue accumulating in history until you use the clear call application stats command or the gateway reloads.

Note

Statistics for an application are automatically cleared if the application is deleted with the no call

application voice command or its script is reloaded with the call application voice load command.

Examples

The following is sample output from the show call application gateway-level command using different keywords:

Router#

show call application active gateway-level

Gateway level statistics for active application sessions:

Sessions w/ stats:

Currently connected incoming PSTN legs:

Currently connected outgoing PSTN legs:

Currently connected incoming VoIP legs:

1

1

0

0

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Currently connected outgoing VoIP legs:

Placecalls in transit:

Handouts in transit:

Pending ASNL subscriptions:

Pending ASNL unsubscriptions:

Prompts playing (non-TTS):

Recordings:

TTS prompts playing:

The table below describes the fields shown in the display.

0

0

0

0

0

0

0

0

Table 23: show call application active gateway-level Field Descriptions

Field

Sessions w/ stats

Currently connected incoming PSTN legs

Currently connected outgoing PSTN legs

Currently connected incoming VoIP legs

Currently connected outgoing VoIP legs

Placecalls in transit

Handouts in transit

Pending ASNL subscriptions

Pending ASNL unsubscriptions

Prompts playing (non-TTS)

Recordings

TTS prompts playing

Description

Number of active application instances that the gauges represent.

Number of active call legs that are incoming from the

PSTN.

Number of active call legs that are outgoing to the

PSTN.

Number of active call legs that are incoming from the

IP network.

Number of active call legs that are outgoing to the IP network.

Number of outgoing calls in progress for all active application instances. The value is decremented by one after the call is either set up or the setup fails.

Number of handoffs in progress for all active application instances. The value is decremented by one after the receiving application either hands back the application or rejects the handoff.

Number of Application Subscribe Notify Layer

(ASNL) subscription requests that are in progress for all active application instances.

Number of ASNL unsubscription requests that are in progress for all active application instances.

Number of recorded prompts being played in all active application instances.

Number of recordings being made in all active application instances.

Number of text-to-speech (TTS) prompts playing in all active application instances.

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Router#

show call application history gateway-level

Gateway level statistics for history application sessions:

Sessions w/ stats: 2

Last reset time:

Statistics:

*Jul 3 15:49:28 PST

Subscriber Service - Call

PSTN VOIP

Incoming Outgoing Incoming Outgoing

Legs setup:

Total legs connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

2

2

0

0

0

0

2

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

0

3

3

Play

3

0

0

0

1

0

0

0

0

Record

0

0

0

0

0

TTS

0

0

For a description of the fields shown with the history keyword, see the table above.

Related Commands

Command call application stats clear call application stats show call application app-level show call application session-level

Description

Enables statistics collection for voice applications.

Clears application-level statistics in history and subtracts the statistics from the gateway-level statistics.

Displays application-level statistics for voice applications.

Displays event logs and statistics for voice application instances.

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show call application interface

To display event logs and statistics for application interfaces, use the show call application interfacecommand in privileged EXEC mode.

show call application interface [summary| {aaa| asr| flash| http| ram| rtsp| smtp| tftp| tts} [server server]

[event-log| info| summary]]

Syntax Description summary aaa asr flash http ram rtsp smtp tftp tts

server server

event-log info

(Optional) Displays a short summary of all interface types or the selected interface.

Authentication, authorization, and accounting (AAA) interface type.

Automatic speech recognition (ASR) interface type.

Flash memory of the Cisco gateway.

Hypertext Transfer Protocol (HTTP) interface type.

Memory of the Cisco gateway.

Real Time Streaming Protocol (RTSP) interface type.

Simple Mail Transfer Protocol (SMTP) interface type.

Trivial File Transfer Protocol (TFTP) interface type.

Text-to-speech (TTS) interface type.

(Optional) Displays event logs or statistics for the specified server.

(Optional) Displays event logs for the selected interface type or server.

(Optional) Displays statistics for the selected interface type or server.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(8)T

Modification

This command was introduced.

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Usage Guidelines

If you use the server keyword, only statistics or event logs for that server display. To display event logs or statistics with this command, you must enable statistics and event logging with the call application interface

event-log and call application interface stats command, respectively. To reset statistic counters to zero and clear the event logs in history, use the clear call application interface command.

Examples

The following is sample output from the show call application interface command using different keywords:

Router#

show call application interface summary

Aggregated statistics for http service:

Stats last reset time *Jul 3 15:24:48 PST

Read requests:

Read successes:

3

0

Read failures:

Read aborts:

Total bytes read:

Write requests:

Write successes:

Write failures:

Write aborts:

3

0

0

0

0

0

0

Total bytes written: 0

Aggregated statistics for tts service:

Stats last reset time *Jul 3 15:24:48 PST

Read requests:

Read successes:

Read failures:

Read aborts:

0

0

0

0

Aggregated statistics for asr service:

Stats last reset time *Jul 3 15:24:48 PST

Read requests: 0

Read successes:

Read failures:

0

0

Read aborts: 0

Aggregated statistics for tftp service:

Stats last reset time *Jul 3 15:24:48 PST

Read requests: 3

Read successes:

Read failures:

2

0

Read aborts:

Total bytes read:

1

145888

Router#

show call application interface tftp summary

Aggregated statistics for tftp service:

Stats last reset time *Jul 3 15:24:48 PST

Read requests: 3

Read successes:

Read failures:

2

0

Read aborts:

Total bytes read:

Server Name

1

145888

Stats Error Count

172.19.139.145

speech-serv

Y

Y

0

0

Router#

show call application interface tftp

Server name: 172.19.139.145

Statistics:

Last reset time *Jul 3 16:08:13 PST

Read requests:

Read successes:

Read failures:

Read aborts:

1

2

0

1

145888 Total bytes read:

Event log:

Last reset time *Jul 3 16:08:13 PST buf_size=50K, log_lvl=INFO

Event Log

Y

N

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<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

172.19.139.145:1057277293:53:INFO: ID = 6549D9E0: Read requested for URL = tftp://172.19.139.145/audio/ch_welcome.au

172.19.139.145:1057277295:54:INFO: ID = 6549D9E0: Streamed read transaction Successful URL

= tftp://172.19.139.145/audio/ch_welcome.au

172.19.139.145:1057277306:59:INFO: ID = 649A0320: Streamed read transaction Successful URL

= tftp://172.19.139.145/audio/ch_welcome.au

172.19.139.145:1057277317:65:INFO: ID = 650922A8: Read request aborted for URL = tftp://172.19.139.145/audio/ch_welcome.au

--------------------------------------------------------------------------------

Router#

show call application interface tftp event-log

Server name:

Event log:

172.19.139.145

Last reset time *Jul 3 16:08:13 PST buf_size=50K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

172.19.139.145:1057277293:53:INFO: ID = 6549D9E0: Read requested for URL = tftp://172.19.139.145/audio/ch_welcome.au

172.19.139.145:1057277295:54:INFO: ID = 6549D9E0: Streamed read transaction Successful URL

= tftp://172.19.139.145/audio/ch_welcome.au

172.19.139.145:1057277306:59:INFO: ID = 649A0320: Streamed read transaction Successful URL

= tftp://172.19.139.145/audio/ch_welcome.au

172.19.139.145:1057277317:65:INFO: ID = 650922A8: Read request aborted for URL = tftp://172.19.139.145/audio/ch_welcome.au

--------------------------------------------------------------------------------

Router#

show call application interface tftp info

Server name:

Statistics:

172.19.139.145

Last reset time *Jul 3 16:08:13 PST

Read requests: 3

Read successes:

Read failures:

2

0

Read aborts:

Total bytes read:

1

145888

--------------------------------------------------------------------------------

The table below describes the significant fields shown in the display.

Table 24: show call application interface Field Descriptions

Field

Last reset time

Read requests

Read successes

Read failures

Read aborts

Total bytes read

Description

Time at which the statistics were last cleared with the

clear call application interface command, or the gateway was restarted.

Total number of read requests from applications to this interface type.

Number of successful read requests from applications to this interface type.

Number of failed read requests from applications to this interface type.

Number of aborted read requests from applications to this interface type.

Total number of bytes that the application read from this interface type.

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Field

Server name

Stats

Error Count

Event Log

Related Commands

Command call application interface event-log call application interface stats clear call application interface show aal2 profile through show call filter match-list

Description

Name of the specific server.

Whether statistics are enabled for this server.

Total number of errors for this server.

Whether event logging is enabled for this server.

Description

Enables event logging for external interfaces used by voice applications.

Enables statistics collection for application interfaces.

Clears application interface statistics and event logs.

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show call application services registry

To display a one-line summary of all TCL IVR 2.0 application sessions that have registered as a service, use the show call application services registrycommand in user EXEC or privileged EXEC mode.

show call application services registry

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

12.3(4)T

Modification

This command was introduced.

Usage Guidelines

Examples

• The services registry is a database that keeps track of every TCL IVR 2.0 application instance that registers as a service. Other TCL applications can then find and communicate with any registered application.

• A TCL session is not registered as a service through a Cisco IOS command. A running instance of a

TCL IVR 2.0 application registers itself as a service with the TCL service register command. For information about the service register command, refer to the TCL IVR API Version 2.0 Programmer’s

Guide

.

The following is sample output for this command:

Router#

show call application services registry

There are 1 Registered Services

Service Name data_service

Session ID Session Name

4 s1

The table below describes significant fields in the display.

Table 25: show call application services registry Field Descriptions

Field

Service Name

Session ID

Description

Name specified by the TCL service register command.

ID of the session that registered as this service. You can use this ID in the show call application sessions

id command to view details about this session.

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Field

Session Name

Description

Name configured by the call application session

start command, if the session was started on the gateway rather than by an incoming call.

Related Commands

Command Description call application session start (global configuration)

Starts a new instance (session) of a TCL application from global configuration mode.

call application session start (privileged EXEC)

Starts a new instance (session) of a TCL application from privileged EXEC mode.

call application session stop

Stops a voice application session that is running.

show call application sessions

Displays summary or detailed information about voice application sessions.

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show aal2 profile through show call filter match-list show call application session-level

show call application session-level

To display event logs and statistics for individual voice application instances, use the show call application

session-levelcommand in privileged EXEC mode.

show call application {active| history} session-level [summary| [app-tag application-name| last [ number ]|

session-id session-id] [event-log| info]]

Syntax Description active history

summary

app-tag application-name

last

number

session-id session-id event-log info

Displays event logs or statistics for active application instances.

Displays event logs or statistics for inactive application instances in the history table.

Displays a summary of each application instance.

Name of a specific voice application. Output displays event logs or statistics for that voice application.

(Optional) Displays event logs or statistics for the most recent instance.

(Optional) Displays event logs or statistics for this number of most recent previous instances.

Identifies a specific application instance. Output displays event logs or statistics for that instance.

(Optional) Displays event logs for application instances.

(Optional) Displays statistics for application instances.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(8)T

Modification

This command was introduced.

Usage Guidelines

• To display event logs or statistics with this command, you must enable event logging and statistics with the call application event-log and call application stats command, respectively.

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Examples

• This command displays gauges and counters that are aggregated per application instance. The values represent an individual instance running on the gateway while statistics collection is enabled.

• The number of records that are included when using the history keyword depends on the settings of the

call application history session max-records and call application history session retain-timer commands.

The following is sample output from the show call application session-level command using different keywords and arguments:

Router#

show call application active session-level summary

SID Application Name

5 generic

6 generic Y 3 Y *Jul 3 15:19:5

Router#

show call application active session-level last

Session Info:

Session id: 6

Stat Err Cnt

Y 6

Log Start Time

Y *Jul 3 15:19:4

Session name:

Application name:

Application URL:

Start time: generic tftp://demo/scripts/master/generic.vxml

*Jul 3 15:19:53 PST

Statistics:

Subscriber Service - Call

Legs setup:

Total legs connected:

Legs currently connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

0

0

0

0

0

1

1

0

0

PSTN VOIP

Incoming Outgoing Incoming Outgoing

1 0 0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media actives:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

Play

4

0

0

0

4

0

0

0

3

0

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

Record

0

0

0

0

0

0

TTS

0

0

0

0

0

0

6:1057274393:472:INFO: Session started for App-type = generic, URL = tftp://demo/scripts/master/generic.vxml

6:1057274393:473:INFO: Incoming Telephony call received, LegID = 10

6:1057274393:474:INFO: LegID = 10: Calling = 4084644753, called = 52927, dial peer = 1

6:1057274393:475:INFO: LegID = 10: Leg State = LEG_INCCONNECTED

6:1057274393:478:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

6:1057274408:517:INFO: Script received event = "error.badfetch"

Router#

show call application active session-level info

Session Info:

Session id:

Session name:

Application name:

Application URL:

Start time:

5 generic tftp://demo/scripts/master/generic.vxml

*Jul 3 15:19:44 PST

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Statistics:

Subscriber Service - Call

Legs setup:

Total legs connected:

Legs currently connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

PSTN VOIP

Incoming Outgoing Incoming Outgoing

1 0 0 0

1

1

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media actives:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

Session Info:

Session id:

Session name:

6

Play

9

0

0

0

9

0

0

0

8

0

Record

0

0

0

0

0

0

TTS

0

0

0

0

0

0

Application name:

Application URL:

Start time:

Statistics: generic tftp://demo/scripts/master/generic.vxml

*Jul

Subscriber Service - Call

3 15:19:53 PST

PSTN VOIP

Incoming Outgoing Incoming Outgoing

Legs setup:

Total legs connected:

Legs currently connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

3

3

1

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

TTS

0

0

0

Media attempts:

Media actives:

Media successes:

Media aborts:

Media failures:

Media duration (in seconds):

Application Internal Service - Handoff

Play

7

0

0

0

7

0

Record

0

0

0

0

0

0

Bridged handoffs:

Bridged handoffs returned:

Blind handoffs:

Handoffs in transit:

Handoffs failed:

Incoming Outgoing

0 0

0

0 x x

0

0

0

0

Application Internal Service - Placecall/transfer

Placecall requests: 0

Placecall successes:

Placecall failures:

0

0

Placecalls in transit: 0

Application Internal Service - Document Read-Write

Doc requests:

Doc successes:

Doc failures:

Read

0

0

0

Application Internal Service - Downloaded Script

Write

0

0

0

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Script parse errors:

Application Internal Service - ASNL

ASNL notifications:

0

ASNL requests:

ASNL successes:

ASNL pendings:

ASNL failures:

Subscriber Interaction - DTMF

0

Subscription Unsubscription

0 0

0

0

0

0

0

0

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

Subscriber Interaction - ASR

ASRs not matched:

ASRs matched:

ASRs no input:

Subscriber Interaction - AAA

0

0

6

0

0

0

0

AAA successes:

Authentication Authorization

0 0

AAA failures: 0 0

Router#

show call application active session-level event-log

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

5:1057274384:454:INFO: Session started for App-type = generic, URL = tftp://demo/scripts/master/generic.vxml

5:1057274384:455:INFO: Incoming Telephony call received, LegID = D

5:1057274384:456:INFO: LegID = D: Calling = 4085550198, called = 52927, dial peer = 1

5:1057274384:457:INFO: LegID = D: Leg State = LEG_INCCONNECTED

5:1057274384:460:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

5:1057274384:462:ERR : Prompt play setup failure.

5:1057274384:463:INFO: Script received event = "error.badfetch"

5:1057274389:464:INFO: Timed out waiting for user DTMF digits, no user input.

5:1057274389:465:INFO: Script received event = "noinput"

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

6:1057274393:472:INFO: Session started for App-type = generic, URL = tftp://demo/scripts/master/generic.vxml

6:1057274393:473:INFO: Incoming Telephony call received, LegID = 10

6:1057274393:474:INFO: LegID = 10: Calling = 4084644753, called = 52927, dial peer = 1

6:1057274393:475:INFO: LegID = 10: Leg State = LEG_INCCONNECTED

6:1057274393:478:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

6:1057274393:480:ERR : Prompt play setup failure.

6:1057274393:481:INFO: Script received event = "error.badfetch"

6:1057274398:488:INFO: Timed out waiting for user DTMF digits, no user input.

6:1057274398:489:INFO: Script received event = "noinput"

6:1057274398:490:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

Router#

show call application active session-level app-tag generic

Session Info:

Session id:

Session name:

5

Application name:

Application URL:

Start time:

Statistics: generic tftp://demo/scripts/master/generic.vxml

*Jul

Subscriber Service - Call

3 15:19:44 PST

PSTN VOIP

Incoming Outgoing Incoming Outgoing

Legs setup:

Total legs connected:

Legs currently connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

1

1

1

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

282

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Media attempts:

Media actives:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

Play

16

0

0

0

17

0

0

0

16

0

Record

0

0

0

0

0

0

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

5:1057274384:454:INFO: Session started for App-type = generic, URL = tftp://demo/scripts/master/generic.vxml

5:1057274384:455:INFO: Incoming Telephony call received, LegID = D

TTS

0

0

0

0

0

0

5:1057274384:456:INFO: LegID = D: Calling = 4085550198, called = 52927, dial peer = 1

5:1057274384:457:INFO: LegID = D: Leg State = LEG_INCCONNECTED

5:1057274384:460:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

5:1057274384:462:ERR : Prompt play setup failure.

5:1057274384:463:INFO: Script received event = "error.badfetch"

5:1057274389:464:INFO: Timed out waiting for user DTMF digits, no user input.

5:1057274389:465:INFO: Script received event = "noinput"

5:1057274389:466:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

Router#

show call application active session-level session-id 7

Session Info:

Session id:

Session name:

Application name:

Application URL:

7 generic tftp://demo/scripts/master/generic.vxml

Start time:

Statistics:

*Jul

Subscriber Service - Call

3 15:21:26 PST

Legs setup:

Total legs connected:

Legs currently connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

1

0

0

0

0

0

PSTN VOIP

Incoming Outgoing Incoming Outgoing

1 0 0 0

1 0 0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media actives:

Media successes:

Media aborts:

Play

3

0

0

0

3

0

Record

0

0

0

0

0

0

Media failures:

Total media duration (in seconds):

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

0

0

2

0 DTMFs long pound:

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

7:1057274486:662:INFO: Session started for App-type = generic, URL =

TTS

0

0

0

0

0

0 tftp://demo/scripts/master/generic.vxml

7:1057274486:663:INFO: Incoming Telephony call received, LegID = 13

7:1057274486:664:INFO: LegID = 13: Calling = 4085550198, called = 52927, dial peer = 1

7:1057274486:665:INFO: LegID = 13: Leg State = LEG_INCCONNECTED

7:1057274486:668:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

Router#

show call application history session-level summary

SID Application Name

1 generic

Stat Err Cnt

Y 3

Log Stop Time Duration

Y *Jul 3 15:49:2 00:00:11

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2 generic Y 1 Y *Jul 3 15:49:3 00:00:03

Router#

show call application history session-level last

Session Info:

Session id:

Session name:

Application name:

Application URL:

Start time:

2 generic tftp://demo/scripts/master/generic.vxml

*Jul 3 15:49:29 PST

*Jul 3 15:49:33 PST Stop time:

Statistics:

Subscriber Service - Call

PSTN VOIP

Incoming Outgoing Incoming Outgoing

Legs setup:

Total legs connected:

Legs handed in:

Legs handed in returned back:

1

1

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

0

0

1

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media successes:

Media aborts:

Play

1

0

0

Record

0

0

0

Media failures:

Total media duration (in seconds):

Event log: buf_size=25K, log_lvl=INFO

1

0

0

0

0

0

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

2:1057276169:28:INFO: Session started for App-type = generic, URL =

TTS

0

0

0 tftp://demo/scripts/master/generic.vxml

2:1057276169:29:INFO: Incoming Telephony call received, LegID = 4

2:1057276169:30:INFO: LegID = 4: Calling = 4085550198, called = 52927, dial peer = 1

2:1057276169:31:INFO: LegID = 4: Leg State = LEG_INCCONNECTED

2:1057276169:34:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

2:1057276169:36:ERR : Prompt play setup failure.

2:1057276169:37:INFO: Script received event = "error.badfetch"

2:1057276173:39:INFO: Script received event = "telephone.disconnect.hangup"

2:1057276173:40:INFO: LegID = 4: Call disconnected, cause = normal call clearing (16)

2:1057276173:43:INFO: Session done, terminating cause =

Router#

show call application history session-level event-log

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

1:1057276157:3:INFO: Session started for App-type = generic, URL = tftp://demo/scripts/master/generic.vxml

1:1057276157:4:INFO: Incoming Telephony call received, LegID = 1

1:1057276157:5:INFO: LegID = 1: Calling = 4085550198, called = 52927, dial peer = 1

1:1057276157:6:INFO: LegID = 1: Leg State = LEG_INCCONNECTED

1:1057276157:9:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

1:1057276160:12:ERR : Prompt play setup failure.

1:1057276160:13:INFO: Script received event = "error.badfetch"

1:1057276165:14:INFO: Timed out waiting for user DTMF digits, no user input.

1:1057276165:15:INFO: Script received event = "noinput"

1:1057276165:16:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

1:1057276165:18:ERR : Prompt play setup failure.

1:1057276165:19:INFO: Script received event = "error.badfetch"

1:1057276168:21:INFO: Script received event = "telephone.disconnect.hangup"

1:1057276168:22:INFO: LegID = 1: Call disconnected, cause = normal call clearing (16)

1:1057276168:25:INFO: Session done, terminating cause =

Event log: buf_size=25K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

2:1057276169:28:INFO: Session started for App-type = generic, URL = tftp://demo/scripts/master/generic.vxml

2:1057276169:29:INFO: Incoming Telephony call received, LegID = 4

2:1057276169:30:INFO: LegID = 4: Calling = 4085550198, called = 52927, dial peer = 1

2:1057276169:31:INFO: LegID = 4: Leg State = LEG_INCCONNECTED

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2:1057276169:34:INFO: Playing prompt #1: http://172.19.139.145/audio/ch_welcome.au

2:1057276169:36:ERR : Prompt play setup failure.

2:1057276169:37:INFO: Script received event = "error.badfetch"

2:1057276173:39:INFO: Script received event = "telephone.disconnect.hangup"

2:1057276173:40:INFO: LegID = 4: Call disconnected, cause = normal call clearing (16)

2:1057276173:43:INFO: Session done, terminating cause =

Router#

show call application history session-level info

Session Info:

Session id: 1

Session name:

Application name:

Application URL:

Start time: generic tftp://demo/scripts/master/generic.vxml

Stop time:

Statistics:

*Jul 3 15:49:17 PST

*Jul

Subscriber Service - Call

3 15:49:28 PST

Legs setup:

Total legs connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

0

0

0

0

1

0

0

PSTN VOIP

Incoming Outgoing Incoming Outgoing

1 0 0 0

1 0 0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

Subscriber Interaction - DTMF

DTMFs not matched:

DTMFs matched:

DTMFs no input:

DTMFs long pound:

Session Info:

Session id:

Session name:

2

Play

2

0

0

2

3

0

0

1

0

Record

0

0

0

0

0

TTS

0

0

0

0

0

Application name:

Application URL:

Start time: generic tftp://demo/scripts/master/generic.vxml

Stop time:

Statistics:

*Jul 3 15:49:29 PST

*Jul

Subscriber Service - Call

3 15:49:33 PST

Legs setup:

Total legs connected:

Legs handed in:

Legs handed in returned back:

Legs handed out:

Legs handed out came back:

Legs disconnected normally:

Legs disconnected for user error:

Legs disconnected for system error:

Subscriber Service - Media

PSTN VOIP

Incoming Outgoing Incoming Outgoing

1 0 0 0

1 0 0 0

0

0

0

0

1

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Media attempts:

Media successes:

Media aborts:

Media failures:

Total media duration (in seconds):

0

1

0

Play

1

0

Record

0

0

0

0

0

TTS

0

0

0

0

0

The table below describes significant fields in the displays.

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Note

These fields display for the show call application session-level, show call application app-level, and

show call application gateway-level commands. At the session level, the fields apply to a single application instance. At the application level, the fields apply to all instances of an application. At the gateway level, the fields apply to all instances of all applications.

Table 26: show call application active session-level info Field Descriptions

Field

Session id

Session name

Application name

Application URL

Description

Session ID assigned to the instance when it became active.

Name of the session defined with the call application

session start command.

Name of the application defined with the call

application voice command.

Location of the application script defined with the

call application voice command.

Time at which the session started.

Start time

Subscriber Service -- Call

Legs setup

Total legs connected

Legs currently connected

Legs handed in

Legs handed in returned back

Legs handed out

Legs handed out came back

Legs disconnected normally

Number of calls setup (indications and requests) by an application instance.

Number of calls connected by an application instance.

Number of calls currently connected by an application instance at any moment.

Number of call legs received as an incoming handoff from another application.

Number of call legs received as an incoming handoff from another application that were returned to the other application.

Number of call legs handed off to another application.

Number of call legs handed off to another application that were returned by the other application.

Number of incoming and outgoing calls disconnected for normal causes.

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Field

Legs disconnected for user error

Legs disconnected for system error

Subscriber Service -- Media

Media attempts

Media actives,

Media successes

Media aborts

Media failures

Total media duration

Application Internal Service -- Handoff

Bridged handoffs, incoming

Bridged handoffs, outgoing

Bridged handoffs returned, incoming

Bridged handoffs returned, outgoing

Blind handoffs, incoming

show call application session-level

Description

Number of incoming calls disconnected for call failure reasons, such as no answer or busy.

Number of incoming calls disconnected for system failure reasons, such as no resources.

Number of prompt playouts, recordings, and text-to-speech (TTS) attempts on call legs in this application instance.

Number of prompt playouts, recordings, and TTS prompts currently active on call legs in an application instance.

Number of prompt playouts, recordings, and TTS prompts that were successful on call legs in an application instance.

Number of prompt playouts, recordings, and TTS prompts that were aborted by the caller on call legs in an application instance.

Number of prompt playouts, recording, and TTS attempts that failed on call legs in an application instance.

Total duration, in seconds, of prompt playing, recording, or TTS.

Number of handoffs received with callback (bridged transfers) in an application instance.

Number of handoffs placed with callback (bridged transfers) by an application instance.

Number of incoming bridged handoffs that were returned by an application instance.

Number of outgoing bridged handoffs that were returned to an application instance.

Number of handoffs received with no callback (blind transfers) in an application instance.

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Field

Blind handoffs, outgoing

Handoffs in transit

Handoffs failed

1

Description

Number of handoffs placed with no callback (blind transfers) by an application instance.

Number of handoffs in progress for an application instance. The value is decremented by one after the receiving application either hands back the application or rejects the handoff.

Number of handoffs that failed (bridged and blind) in an application instance.

Application Internal Service -- Placecall/transfer

Placecall requests

Placecall successes

Placecall failures

Placecalls in transit show call application session-level, on page 279

Number of outgoing call setup requests made by an application instance.

Number of outgoing calls placed by an application instance.

Number of outgoing call setup requests that failed for an application instance.

Number of outgoing calls in progress for an application. The value is decremented by one after the call is either set up or the setup fails.

Application Internal Service -- Document Read-Write

Doc requests

Doc successes

Doc failures

Number of document fetch and submit requests.

Number of successful document fetches and submits.

Number of document fetch and submit failures.

Application Internal Service -- Downloaded Script

Script parse errors Number of semantic errors seen by an application instance.

Application Internal Service -- ASNL

ASNL notifications Number of Application Subscribe Notify Layer

(ASNL) notifications received from servers.

ASNL requests

ASNL successes

Number of subscribe or unsubscribe requests made by an application instance.

Number of subscribe or unsubscribe requests that succeeded for an application instance.

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Field

ANSL failures

Description

Number of subscribe or unsubscribe requests that failed for an application instance.

Subscriber Interaction --- DTMF

DTMFs not matched

DTMFs matched

DTMFs no input

DTMFs long pound

Number of DTMF patterns input by a caller that were not matched in an application instance.

Number of DTMF patterns input by a caller that were matched in an application instance.

Number of "no input" notifications received (includes

DTMF timeouts).

Number of long-pound interrupts from a caller seen by an application instance.

Subscriber Interaction -- ASR

ASR not matched

ASR matched

ASR no inputs

Number of automatic speech recognition (ASR) phrases from a caller that were not matched in an application instance.

Number of automatic speech recognition (ASR) phrases from a caller that were matched in an application instance.

Number of "no input" notifications received from

ASR servers.

Subscriber Interaction -- AAA Authentication

AAA authentication successes

AAA authentication failures

Number of AAA authentication successes.

Number of AAA authentication failures because of invalid passwords.

Subscriber Interaction -- AAA Authorizations

AAA authorization successes

AAA authorization failures

Number of AAA authorization successes.

Number of AAA authorization failures.

1 When this gauge is greater than zero, the application instance might stop processing the script and the counters and gauges may appear to freeze. When the handoff or the placecall operation is finished and control is returned to the application instance, the counters and gauges are updated.

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Related Commands

Command call application event-log call application history session max-records call application history session retain-timer call application stats call application voice event-log show call application app-level show call application gateway-level

Description

Enables event logging for voice application instances.

Sets the maximum number of application instance records saved in history.

Sets the maximum number of minutes for which application instance records are saved in history.

Enables statistics collection for voice applications.

Enables event logging for a specific voice application.

Displays application-level statistics for voice applications.

Displays gateway-level statistics for voice application instances.

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show call application sessions

To display summary or detailed information about all running or stopped voice application sessions, use the

show call application sessions command in user EXEC or privileged EXEC mode.

show call application sessions [callid call-id| id session-id| name instance-name]

Syntax Description

callid call-id

id session-id

name instance-name

(Optional) Call-leg ID of an active call that is being controlled by the session.

(Optional) Session ID for the specific application instance.

(Optional) Name assigned to the instance with the

call application session start command.

Command Default

No default behavior or values

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

12.3(4)T

Modification

This command was introduced.

Usage Guidelines

• A specific application session is identified by one of three different methods: call ID, session ID, or instance name.

• If a specific session is identified by a callid, id, or name keyword, this command displays information about that specific session only. If you do not use a keyword, this command displays a one-line summary of all sessions, not just those sessions that are started by the call application session start command.

• This command lists all running TCL IVR 2.0 and VoiceXML application sessions and TCL sessions that are stopped. A session displays a state of "stopped" if you intentionally stop it with the call

application session stop or no call application session start command, or because there is a syntax error that prevents the script from running. This is the case only if the session is started with the call

application session start command through global configuration mode.

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Examples

Note

If a session is started with the call application session start command in privileged EXEC mode, it is not tracked by the system and is therefore not shown as stopped in the output of the show call application

sessions command.

The following is sample output from this command:

Router#

show call application sessions

TCL Sessions

There are 1 active TCL sessions

SID Name

5 serv1

Called Calling

VXML Sessions

No running VXML sessions

Stopped Sessions

Instance Name my_instance1

App Name sample

State stopped

The table below describes significant fields in the display.

Table 27: show call application sessions Field Descriptions

Field

SID

Name

Called

Calling

App Name

Legs

State

App Name sample_service

Legs

Description

Session identifier for active sessions.

Session name that was configured with the call

application session start command.

Called number for active calls that are using the session.

Calling number for active calls that are using the session.

Name of the application for which the instance was created.

Any active call legs that are controlled by the session.

Shows "stopped" for any session that is no longer running, provided that the session is started with the

call application session start command in global configuration mode.

The following is sample output for a session named serv1:

Router#

show call aplication sessions name serv1

Session named serv1 is in the start list in state running

It is configured to start on GW reboot

The application it runs is sample_service

Handle is TCL_HAND*1653710732*0*3193204

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TCL Session ID B

App: sample_service

URL: tftp://dev/demo/scripts/sample_service.tcl

Session name: serv1

Session handle: TCL_HAND*1653710732*0*3193204

FSM State: start_state

ID for 'show call active voice id' display: 0

Legs:

Services: data_service

The table below describes significant fields in the display.

Table 28: show call application sessions name Field Descriptions

Field

App

URL

Session name

Session handle

FSM State

ID for ‘show call active voice id’ display:

Legs

Services

Description

Name of the application for which the instance was created.

Location of the script used for the application as specified with the call application voice command.

Session name that was configured with the call

application session start command.

Handle that is returned from the TCL mod_handle infotag. A session handle is used in a TCL script on a Cisco gateway to send messages to other sessions.

Current state in the TCL IVR 2.0 finite-state machine, as specified with the TCL fsm setstate command in the script.

Call identifier.

Any active call legs that are controlled by this session.

Service name for the session if it registered as a service with the TCL service register command in the script. You can display a list of all registered services with the show call application services registry command.

Related Commands

Command Description call application session start (global configuration)

Starts a new instance (session) of a TCL application from global configuration mode.

call application session start (privileged EXEC)

Starts a new instance of a TCL application from privileged EXEC mode.

call application session stop

Stops a voice application session that is running.

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Command show call application services registry

Description

Displays a one-line summary of all registered services.

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show call application voice

To display information about voice applications, use the show call application voice command in EXEC mode.

show call application voice [name| summary]

Syntax Description

name

summary

(Optional) Name of the desired voice application.

Output displays information about that application.

(Optional) Output displays a one-line summary of each voice application.

Command Default

If both the name argument and summary keyword are omitted, command output displays detailed information about all interactive voice response (IVR) applications.

Command Modes

EXEC (#)

Command History

Release

11.3(6)NA2

12.0(3)T

12.1(5)T

12.1(5)XM2

12.2(2)XB

12.2(2)XB1

12.2(4)XM

12.2(8)T

12.2(11)T

12.3(14)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.0(3)T.

This command was implemented on the Cisco AS5800.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was modified to support VoiceXML applications.

This command was implemented on the Cisco AS5850.

This command was implemented on the Cisco 1750 and Cisco 1751. This command was not supported on any other platforms in this release.

This command was implemented on the Cisco 1751, Cisco 2600 series, Cisco

3600 series, Cisco 3725, Cisco 3745, and Cisco 7200.

This command was integrated into Cisco IOS Release 12.2(11)T for VoiceXML applications. This command is supported on the Cisco AS5300, Cisco AS5350,

Cisco AS5400, and Cisco AS5850 in this release.

New output was added relating to the SCCP Telephony Control Application

(STCAPP).

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Usage Guidelines

The show call application voicecommand displays a detailed description of each configured application.

If the name of a specific application is entered, the command displays detailed information about only that application.

If the summary keyword is entered, the command displays a one-line summary about each application.

If STCAPP is enabled, the summary command displays STCAPP as an available call application.

If an asterisk is displayed next to the application name when the summary keyword is used, the application is configured, but not running. Normally this is because the application was not successfully loaded, for example: name

*vapptest2 description flash:helloworld.vxml

TCL scripts and VoiceXML documents can be stored in any of the following locations: TFTP, FTP, or HTTP servers; Flash memory of the gateway; or the removable disks of the Cisco 3600 series. The audio files that they use can be stored in any of these locations and on RTSP servers.

Examples

The following example shows the output for the session Toolkit Command Language (TCL) script:

Router#

show call application voice session

Application session

The script is compiled into the image

It has 0 calls active.

Interpreted by infrastructure version 2.0

The TCL Script is:

------------------

# app_session.tcl

#----------------------------------

# August 1999, Saravanan Shanmugham

#

# Copyright (c) 1998, 1999, 2000, 2001 by cisco Systems, Inc.

# All rights reserved.

#----------------------------------

#

# This tcl script mimics the default SESSION app

#

# If DID is configured, just place the call to the dnis

# Otherwise, output dial-tone and collect digits from the

# caller against the dial-plan.

#

# Then place the call. If successful, connect it up, otherwise

# the caller should hear a busy or congested signal.

# The main routine just establishes the statemachine and then exits.

# From then on the system drives the statemachine depending on the

# events it recieves and calls the appropriate tcl procedure

#---------------------------------

# Example Script

#--------------------------------proc init { } { global param set param(interruptPrompt) true set param(abortKey) * set param(terminationKey) #

} proc act_Setup { } { global dest global beep set beep 0 if { [infotag get leg_isdid] } { set dest [infotag get leg_dnis]

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leg proceeding leg_incoming leg setup $dest callInfo leg_incoming fsm setstate PLACECALL

} else { leg setupack leg_incoming playtone leg_incoming tn_dial set param(dialPlan) true leg collectdigits leg_incoming param

}

} proc act_GotDest { } { global dest set status [infotag get evt_status] if { $status == "cd_004" } { set dest [infotag get evt_dcdigits] leg proceeding leg_incoming leg setup $dest callInfo leg_incoming

} else { puts "\nCall [infotag get con_all] got event $status collecting destina" call close

}

} proc act_CallSetupDone { } { global beep set status [infotag get evt_status] if { $status == "ls_000"} { set creditTimeLeft [infotag get leg_settlement_time leg_all] if { ($creditTimeLeft == "unlimited") ||

($creditTimeLeft == "uninitialized") } { puts "\n Unlimited Time"

} else {

# start the timer for ...

if { $creditTimeLeft < 10 } { set beep 1 set delay $creditTimeLeft

} else { set delay [expr $creditTimeLeft - 10]

}

} else {

} timer start leg_timer $delay leg_incoming puts "Call [infotag get con_all] got event $status collecting destinati" call close

}

} proc act_Timer { } { global beep global incoming global outgoing set incoming [infotag get leg_incoming] set outgoing [infotag get leg_outgoing] if { $beep == 0 } {

#insert a beep ...to the caller connection destroy con_all set beep 1

} else { connection destroy con_all fsm setstate LASTWARN

}

} proc act_LastWarn { } { media play leg_incoming flash:out_of_time.au

} proc act_Destroy { } { media play leg_incoming flash:beep.au

} proc act_Beeped { } { global incoming global outgoing connection create $incoming $outgoing

} proc act_ConnectedAgain { } { timer start leg_timer 10 leg_incoming

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} proc act_Ignore { } {

# Dummy puts "Event Capture"

} proc act_Cleanup { } { call close

} init

#----------------------------------

# State Machine

#-----------------set fsm(any_state,ev_disconnected) "act_Cleanup set fsm(CALL_INIT,ev_setup_indication) "act_Setup set fsm(GETDEST,ev_collectdigits_done) "act_GotDest set fsm(PLACECALL,ev_setup_done) "act_CallSetupDone set fsm(CALLACTIVE,ev_leg_timer) "act_Timer set fsm(INSERTBEEP,ev_destroy_done) "act_Destroy same_state"

GETDEST"

PLACECALL"

CALLACTIVE"

INSERTBEEP" same_state" set fsm(INSERTBEEP,ev_media_done) "act_Beeped set fsm(INSERTBEEP,ev_create_done) "act_ConnectedAgain same_state"

CALLACTIVE" set fsm(LASTWARN,ev_destroy_done) "act_LastWarn set fsm(CALLACTIVE,ev_disconnected) "act_Cleanup set fsm(CALLDISCONNECT,ev_disconnected) "act_Cleanup

CALLDISCONNECT"

CALLDISCONNECT" same_state" set fsm(CALLDISCONNECT,ev_media_done) "act_Cleanup set fsm(CALLDISCONNECT,ev_disconnect_done) "act_Cleanup set fsm(CALLDISCONNECT,ev_leg_timer) fsm define fsm CALL_INIT

"act_Cleanup same_state" same_state" same_state"

The following is sample output for the summary keyword:

Router#

show call application voice summary

name description session fax_hop_on clid_authen

Basic app to do DID, or supply dialtone.

Script to talk to a fax redialer

Authenticate with (ani, dnis) clid_authen_collect Authenticate with (ani, dnis), collect if that fails clid_authen_npw Authenticate with (ani, NULL) clid_authen_col_npw Authenticate with (ani, NULL), collect if that fails clid_col_npw_3 clid_col_npw_npw

Authenticate with (ani, NULL), and 3 tries collecting

Authenticate with (ani, NULL) and 3 tries without pw

DEFAULT lib_off_app

Default system session application

Libretto Offramp

TCL Script Version 2.0 supported.

TCL Script Version 1.1 supported.

Voice Browser Version 2.0 for VoiceXML 1.0 & 2.0 supported.

The following is sample output for the summary keyword when STCAPP is enabled:

Router#

show call application voice summary

SERVICES (standalone applications): name ipsla-responder type

Tcl Script clid_authen clid_col_npw_npw

Tcl Script

Tcl Script

DEFAULT

CTAPP clid_authen_col_npw fax_hop_on ipsla-testcall clid_authen_npw session clid_authen_collect clid_col_npw_3 lib_off_app

DEFAULT.C.OLD

stcapp

MGCPAPP

C Script

C Script

Tcl Script

Tcl Script

Tcl Script

Tcl Script

Tcl Script

Tcl Script

Tcl Script

CCAPI

CCAPI

CCAPI

CCAPI description builtin:app_test_rcvr_script.tcl

builtin:app_clid_authen_script.tcl

builtin:app_clid_col_npw_npw_script.tcl

builtin:Session_Service.C

builtin:CallTreatment_Service.C

builtin:app_clid_authen_col_npw_script.tcl

builtin:app_fax_hop_on_script.tcl

builtin:app_test_place_script.tcl

builtin:app_clid_authen_npw_script.tcl

builtin:app_session_script.tcl

builtin:app_clid_authen_collect_script.tcl

builtin:app_clid_col_npw_3_script.tcl

Libretto Offramp

Obsolete system session application

SCCP Call Control Application

MGCP Application

The following is sample output for the stcapp keyword when the STCAPP is enabled:

Router#

show call application voice stcapp

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App Status:

CCM Status:

CCM Group:

Registration Mode:

Total Devices:

Active

UP

2

CCM

5

Total Calls in Progress: 0

Total Call Legs in Use: 0

The following is sample output from the show call application voice command for a VoiceXML application named vapptest1:

Router#

show call application voice vapptest1

VXML Application vapptest1

URL=flash:demo0.vxml

Security not trusted

No languages configured

It has: 0 calls active.

0 incoming calls

0 calls handed off to it

0 call transfers initiated

0 pages loaded, 0 successful

0 prompts played

0 recorded messages

Interpreted by Voice Browser Version 2.0 for VoiceXML 1.0 & 2.0.

The VXML Script is:

------------------

<?xml version="1.0"?>

<vxml version="1.0">

<form>

<block>

<audio src="flash:demo0.au"/>

</block>

</form>

</vxml>

The table below describes the fields shown in the show call application voice display:

Table 29: show call application voice Field Descriptions

Field

URL

It has: n calls active.

incoming calls calls handed off to it call transfers initiated pages loaded successful

Description

Location of the document used by the application.

Number of calls that are using this application.

Number of incoming public switched telephone network (PSTN) or IP calls that invoked this application.

Number of calls that were handed off to this application by another TCL or VoiceXML application.

Number of call transfers that were initiated by this application.

Number of VoiceXML pages that were loaded by the application.

Number of VoiceXML pages that were completed.

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Field

prompts played recorded messages

Interpreted by

The TCL or VoiceXML Script is

Related Commands

Command call application voice call application voice load show aal2 profile through show call filter match-list

Description

Number of audio prompts that were played by the application.

Number of audio recordings made by the VoiceXML application.

Programming language used by the application.

Content of the VoiceXML document or TCL script.

Description

Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with the application.

Reloads the designated TCL script or VoiceXML document.

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show call fallback cache

To display the current Calculated Planning Impairment Factor (ICPIF) estimates for all IP addresses in cache, use the show call fallback cache command in EXEC mode.

show call fallback cache [ ip-address ]

Syntax Description

ip -address (Optional) Specific IP address.

Command Modes

EXEC (#)

Command History

Release

12.1(3)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Usage Guidelines

Use this command to clear all entries in the cache.

Examples

The following example displays output from this command:

Router#

show call fallback cache

Probe IP Address

--------------

1

2

1.1.1.4

122.24.56.25

2 active probes

Codec

----g729r8

Delay

-----

40 g729r8 148

Loss

----

0

10

Field

-------

Description

------------

0

ICPIF Reject Accept

---------------

5

0

1

9

4

Probe

IP Address

Codec

Probe number

IP Address to which the probe is sent

Codec Type of the probe

Delay

Loss

ICPIF

Reject

Delay in milliseconds that the probe incurred

Loss in % that the probe incurred

Computed ICPIF value for the probe

Number of times that calls of Codec Type <Codec> were rejected to the IP Address

Number of times that calls of Codec Type <Codec> Accept active probes were accepted to the IP Address

Number of destinations being probed

Router#

show call fallback cache 10.14.115.53

Probe IP Address

--------------

1 10.14.115.53

1 active probes

Codec

----g729r8

ICPIF

-----

0

Reject

------

0

Accept

------

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Field descriptions should be self-explanatory.

Related Commands

Command show call fallback stats

Description

Displays call fallback statistics.

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show call fallback config

To display the call fallback configuration, use the show call fallback config command in EXEC mode.

show call fallback config

Syntax Description

This command has no arguments or keywords.

Command Modes

EXEC (#)

Command History

Release

12.1(3)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Examples

The following example displays output from the show call fallback config command:

Router#

show call fallback config

VoIP fallback config:

Fallback is ON

Using ICPIF threshold:

ICPIF value timeout:20 seconds

ICPIF threshold:20

Number of packets in a probe:20

IP precedence of probe packets:2

Fallback cache size:2 entries

Fallback cache timeout:240 seconds

Instantaneous value weight:65

MD5 Keychain:secret

The table below describes the fields shown in the show call fallback config display.

Table 30: show call fallback config Field Descriptions

Field

Fallback is

Using ICPIF threshold

ICPIF value timeout

ICPIF threshold

Number of packets in a probe

Description

Lists enabled/disabled state of call fallback.

ICPIF is configured to determine network traffic.

Lists probe timeout for collecting ICPIF information.

Lists configured ICPIF threshold.

Lists number of configured packets per probe.

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Field

IP precedence of probe packets

Fallback cache size

Fallback cache timeout

Instantaneous value weight

MD5 Keychain

Description

Lists configured IP precedence for probes.

Number of allowed entries in call fallback cache.

Length of cache timeout, in seconds.

Lists weight configured for calculating cache entry based on new probe and last entry.

MD5 authentication has been configured with a keychain of secret.

Related Commands

Command call fallback monitor show voice trunk-conditioning signaling

Description

Enables the monitoring of destinations without fallback to alternate dial peers.

Enables fallback to alternate dial peers in case of network congestion.

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show call fallback stats

To display the call fallback statistics, use the show call fallback stats command in EXEC mode.

show call fallback stats

Syntax Description

This command has no arguments or keywords.

Command Modes

EXEC (#)

Command History

Release

12.1(3)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced on the Cisco 2600, Cisco 3600, and Cisco

MC3810.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Usage Guidelines

To remove all values, use the clear call fallback stats command.

Examples

The following example displays output from the show call fallback stats command:

Router#

show call fallback stats

VOIP Fallback Stats:

Total accepted calls:3

Total rejected calls:1

Total cache overflows:1

Field

-------

Description

------------

Total accepted calls

Total rejected calls

Total cache overflows

Number of times that calls were successful over IP.

Number of times that calls were rejected over IP.

Number of times that the fallback cache overflowed and required pruning.

The table below describes the fields shown in the

show call fallback stats

display

Table 31: show call fallback stats Fields with Descriptions

Field

Total accepted calls

Total rejected calls

Total cache overflows

Description

Number of times that calls were successful over IP.

Number of times that calls were rejected over IP.

Number of times that the fallback cache overflowed and required pruning.

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Related Commands

Command clear call fallback stats show call fallback cache show aal2 profile through show call filter match-list

Description

Clears the call fallback statistics.

Displays the current ICPIF estimates for all IP addresses in the cache.

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show call filter components

To display the components used for filtering calls, use the show call filter components command in privileged

EXEC mode.

show call filter components

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(4)T

Cisco IOS XE Release 3.10S

Modification

This command was introduced.

This command was integrated into Cisco IOS XE Release 3.10S.

Examples

The following example shows the output from running the show call filter components command. The GCFM is the generic call filter module, which is the internal module that controls which components are filtered:

Router#

show call filter components

The following components registered in GCFM:

ISDN

VTSP

CCAPI

TGRM

DIAL-PEER

NUMBER-TRANSLATION

SSAPP

VOICE-IVR-V2

H323

SIP

CRM

TSP

FAX-FOIP

FAX-FMSP

FAX-DMSP

FAX-MSPI

FAX-MTA

DSPAPI

MGCP

DSMP

H221

The table below describes the significant fields shown in the display.

Table 32: show call filter components Field Descriptions

Field

The following components registered in GCFM:

Description

Shows which components are filtered in the generic call filter module.

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Related Commands

Command call filter match-list voice debug call filter inout debug condition match-list outgoing port show call filter match-list

Description

Create a call filter match list for debugging voice calls.

Display the debug trace inside the GCFM.

Run a filtered debug on a voice call.

Configure debug filtering for the outgoing port.

Display call filter match lists.

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show call filter match-list

To display call filter match lists, use the show call filter match-list command in privileged EXEC mode.

show call filter match-list tag

Syntax Description

tag

Numeric label that uniquely identifies the match list.

Command Default

No default behavior or values.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(4)T

Cisco IOS XE Release 3.10S

Examples

Modification

This command was introduced.

This command was integrated into Cisco IOS XE Release 3.10S.

The following example shows an output from the show call filter match-list command:

Router#

show call filter match-list

********************************************* call filter match-list 9 voice

********************************************* incoming calling-number 50200 incoming called-number 50201 incoming signal local ipv4 10.0.101.22

incoming signal remote ipv4 10.0.101.21

incoming media local ipv4 10.0.101.22

incoming media remote ipv4 10.0.101.21

incoming dialpeer 502 outgoing calling-number 50200 outgoing called-number 50201 outgoing port 6/0:D outgoing dialpeer 501 debug condition match-list is set to EXACT_MATCH

********************************************* call filter match-list 10 voice

********************************************* incoming calling-number 50300 incoming called-number 50301 incoming signal local ipv4 10.0.101.22

incoming signal remote ipv4 10.0.101.21

incoming media local ipv4 10.0.101.22

incoming media remote ipv4 10.0.101.21

incoming dialpeer 504 outgoing calling-number 50300 outgoing called-number 50301

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outgoing port 6/1:D outgoing dialpeer 503 debug condition match-list is set to EXACT_MATCH

The table below describes the significant fields shown in the display.

Table 33: show call filter match-list Field Descriptions

Field

call filter match-list 9 voice

Description

Shows which match list is being displayed.

debug condition match-list is set to EXACT_MATCH Shows whether the debug condition is set for exact match or partial match.

Related Commands

Command call filter match-list voice debug call filter inout debug condition match-list show call filter components

Description

Create a call filter match list for debugging voice calls.

Display the debug trace inside the GCFM.

Run a filtered debug on a voice call.

Display the components used for filtering calls.

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show call history fax, page 313

show call history media, page 325

show call history video, page 334

show call history video record, page 336

show call history voice, page 337

show call language voice, page 344

show call leg, page 346

show call media forking, page 351

show callmon, page 352

show call prompt-mem-usage, page 354

show call resource voice stats, page 358

show call resource voice threshold, page 360

show call rsvp-sync conf, page 362

show call rsvp-sync stats, page 364

show call spike status, page 366

show call threshold, page 368

show call treatment, page 371

show call-router routes, page 374

show call-router status, page 376

show ccm-manager, page 379

show cdapi, page 390

show ces clock-select, page 392

show connect, page 393

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show controllers rs366, page 396

show controllers timeslots, page 398

show controllers voice, page 400

show crm, page 403

show csm, page 406

show csm call, page 415

show cube debug category codes, page 417

show cube status, page 418

show debug condition, page 420

show call history fax through show debug condition

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show call history fax

To display the call history table for fax transmissions, use the show call history fax command in user EXEC or privileged EXEC mode.

show call history fax [brief [id identifier]| compact [duration {less| more} time]| id identifier| last number]

Syntax Description brief

id identifier

compact

duration time

last number

(Optional) Displays a truncated version of the call history table.

(Optional) Displays only the call with the specified identifier. Range is a hex value from 1 to FFFF.

(Optional) Displays a compact version.

(Optional) Displays history information for calls that are longer or shorter than a specified time value. The arguments and keywords are as follows:

• less--Displays calls shorter than the value in the

time argument.

• more--Displays calls longer than the value in the time argument.

time --Elapsed time, in seconds. Range is from

1 to 2147483647.

(Optional) Displays the last calls connected, where the number of calls that appear is defined by the

numberargument. Range is from 1 to100 .

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

11.3(1)T

12.0(3)XG

12.0(4)XJ

12.0(4)T

Modification

This command was introduced on the Cisco 3600 series.

This command was implemented for Voice over Frame Relay (VoFR) on the

Cisco 2600 series and Cisco 3600 series.

This command was modified for store-and-forward fax.

This command was modified. The brief keyword was added, and the command was implemented on the Cisco 7200 series.

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Release

12.0(7)XK

12.1(2)T

12.1(5)XM

12.1(5)XM2

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(8)T

12.2(11)T

12.3(1)

12.3(14)T

12.4(15)T

12.4(16)

12.4(22)T

Modification

This command was modified. The brief keyword was implemented on the Cisco

MC3810.

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was implemented on the Cisco AS5800.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was modified. The output of this command was modified to indicate whether the call in question has been established using Annex E.

This command was integrated into Cisco IOS Release 12.2(4)T.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support for the

Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 was not included in this release.

This command was implemented on the Cisco AS5350, Cisco AS5400, Cisco

AS5800, and Cisco AS5850.

This command was modified. The following fields were added:

FaxRelayMaxJitterBufDepth, FaxRelayJitterBufOverFlow,

FaxRelayHSmodulation, and FaxRelayNumberOfPages.

This command was modified. T.38 fax relay call statistics were made available to Call Detail Records (CDRs) through vendor-specific attributes (VSAs) and added to the call log.

This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

This command was modified. Command output was updated to show IPv6 information.

Usage Guidelines

This command displays a call-history table that contains a list of fax calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number from 0 to 500 using the dial-control-mib command in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed, also specified by the dial-control-mib command. The default timer value is 15 minutes.

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Examples show call history fax

You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the number argument.

To display a truncated version of the call history table, use the brief keyword.

This command applies to both on-ramp and off-ramp store-and-forward fax functions.

The following is sample output from the show call history fax command:

Router#

show call history fax

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 0

MGCP call-legs: 0

Total call-legs: 1

GENERIC:

SetupTime=590180 ms

Index=2

PeerAddress=4085452930

PeerSubAddress=

PeerId=81

PeerIfIndex=221

LogicalIfIndex=145

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=59389

DisconnectTime=68204

CallDuration=00:01:28

CallOrigin=2

ReleaseSource=1

ChargedUnits=0

InfoType=fax

TransmitPackets=295

TransmitBytes=5292

ReceivePackets=2967

ReceiveBytes=82110

TELE:

ConnectionId=[0xD9ACDFF1 0x9F5D11D7 0x8002CF18 0xB9C3632]

IncomingConnectionId=[0xD9ACDFF1 0x9F5D11D7 0x8002CF18 0xB9C3632]

CallID=2

Port=3/0/0 (2)

BearerChannel=3/0/0.1

TxDuration=28960 ms

VoiceTxDuration=0 ms

FaxTxDuration=28960 ms

FaxRate=voice bps

FaxRelayMaxJitterBufDepth = 0 ms

FaxRelayJitterBufOverFlow = 0

FaxRelayHSmodulation = 0

FaxRelayNumberOfPages = 0

NoiseLevel=-120

ACOMLevel=127

SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=4085550130

OriginalCallingOctet=0x0

OriginalCalledNumber=52930

OriginalCalledOctet=0xE9

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=4085550130

TranslatedCallingOctet=0x0

TranslatedCalledNumber=52930

TranslatedCalledOctet=0xE9

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

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GwReceivedCalledNumber=52930

GwReceivedCalledOctet3=0xE9

GwReceivedCallingNumber=4085550130

GwReceivedCallingOctet3=0x0

GwReceivedCallingOctet3a=0x80

The table below provides an alphabetical listing of the fields displayed in the output of the show call history

fax command and a description of each field.

Table 34: show call history fax Field Descriptions

Field

ACOM Level

BearerChannel

Buffer Drain Events

Buffer Fill Events

CallDuration

CallerName

CallOrigin

CallState

ChargedUnits

CodecBytes

CoderTypeRate

ConnectionId

ConnectTime

Consecutive-packets-lost Events

Description

Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler, which is the sum of the Echo Return Loss, Echo Return Loss

Enhancement, and nonlinear processing loss for the call.

Identification of the bearer channel carrying the call.

Total number of jitter buffer drain events.

Total number of jitter buffer fill events.

Length of the call, in hours, minutes, and seconds, hh:mm:ss.

Voice port station name string.

Call origin: answer or originate.

Current state of the call.

Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.

Payload size, in bytes, for the codec used.

Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.

Global call identifier for this gateway call.

Time, in milliseconds (ms), at which the call was connected.

Total number of consecutive (two or more) packet-loss events.

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Field

Corrected packet-loss Events

Dial-Peer

DisconnectCause

DisconnectText

DisconnectTime

EchoCancellerMaxReflector=64

ERLLevel

FaxTxDuration

FaxRelayJitterBufOverFlow

FaxRelayMaxJitterBufDepth

FaxRelayHSmodulation

FaxRelayNumberOfPages

GapFillWithInterpolation

GapFillWithRedundancy

show call history fax

Description

Total number of packet-loss events that were corrected using the RFC 2198 method.

Tag of the dial peer sending this call.

Cause code for the reason this call was disconnected.

Descriptive text explaining the reason for the disconnect.

Time, in ms, when this call was disconnected.

The location of the largest reflector, in ms. The reflector size does not exceed the configured echo path capacity. For example, if 32 ms is configured, the reflector does not report beyond 32 ms.

Current Echo Return Loss (ERL) level for this call.

Duration of fax transmission from this peer to the voice gateway for this call. You can derive the Fax

Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

Count of number of network jitter buffer overflows

(number of packets). These packets are equivalent to lost packets.

Maximum depth of jitter buffer (in ms).

Most recent high-speed modulation used.

Number of pages transmitted.

Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

Duration of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.

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show call history fax

Field

GapFillWithPrediction

GapFillWithSilence

GENERIC

GwReceivedCalledNumber,

GwReceivedCalledOctet3,

GwReceivedCallingNumber,

GwReceivedCallingOctet3,

GwReceivedCallingOctet3a

H323 call-legs

HiWaterPlayoutDelay

ImgPages

Incoming ConnectionId

Index

InfoActivity

InfoType

InSignalLevel

Last Buffer Drain/Fill Event

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Description

Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call.

Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1

compression algorithms.

Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.

Generic or common parameters, that is, parameters that are common for VoIP and telephony call legs.

Call information received at the gateway.

Total H.323 call legs for which call records are available.

High-water-mark Voice Playout FIFO Delay during this call.

The fax pages that have been processed.

The incoming_GUID. It can be different with

ConnectionId (GUID) when there is a long_pound or blast_call feature involved. In those cases, incoming_GUID is unique for all the subcalls that have been generated, and GUID is different for each subcall.

Dial peer identification number.

Active information transfer activity state for this call.

Information type for this call; for example, voice or fax.

Active input signal level from the telephony interface used by this call.

Elapsed time since the last jitter buffer drain or fill event, in seconds.

show call history fax through show debug condition show call history fax

Field

Local UUID

LogicalIfIndex

LoWaterPlayoutDelay

LowerIFName

Media

Modem passthrough signaling method in use

NoiseLevel

OnTimeRvPlayout

OriginalCallingNumber, OriginalCalling Octet,

OriginalCalledNumber, OriginalCalledOctet,

OriginalRedirectCalledNumber,

OriginalRedirectCalledOctet

OutSignalLevel

PeerAddress

Description

Unique identifier generated from the originating user agent.

Index number of the logical interface for this call.

Low-water-mark Voice Playout FIFO Delay during this call.

Physical lower interface information. Appears only if the medium is ATM, Frame Relay (FR), or

High-Level Data Link Control (HDLC).

Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is

TELE. If the call is carried over the voice network side, the entry is either ATM, FR, or HDLC.

Indicates that this is a modem pass-through call and that named signaling events (NSEs)--a

Cisco-proprietary version of named telephone events in RFC 2833--are used for signaling codec upspeed.

The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. This means that you might move to a faster codec when you have both voice and data calls and then slow down when there is only voice traffic.

Active noise level for this call.

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Original call information regarding calling, called, and redirect numbers, as well as octet-3s. Octet-3s are information elements (IEs) of Q.931 that include type of number, numbering plan indicator, presentation indicator, and redirect reason information.

Active output signal level to the telephony interface used by this call.

Destination pattern or number associated with this peer.

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show call history fax

Field

PeerId

PeerIfIndex

PeerSubAddress

Percent Packet Loss

Port

ReceiveBytes

ReceiveDelay

ReceivePackets

ReleaseSource

RemoteIPAddress

RemoteUDPPort

Remote UUID

RoundTripDelay

SelectedQoS

SessionProtocol

SessionTarget

SetupTime

SignalingType

320

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Description

ID value of the peer table entry to which this call was made.

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

Subaddress when this call is connected.

Total percent packet loss.

Identification of the voice port carrying the call.

Number of bytes received by the peer during this call.

Average Playout FIFO Delay plus the Decoder Delay during this voice call.

Number of packets received by this peer during this call.

Number value of the release source.

Remote system IP address for the VoIP call.

Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.

Unique identifier generated from the terminating user agent.

Voice packet round-trip delay between the local and remote systems on the IP backbone for this call.

Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.

Session protocol used for an Internet call between the local and remote routers through the IP backbone.

Session target of the peer used for this call.

Value of the system UpTime, in ms, when the call associated with this entry was started.

Signaling type for this call; for example, channel-associated signaling (CAS) or common-channel signaling (CCS).

show call history fax through show debug condition show call history fax

Field

SIP call-legs

Telephony call-legs

Time between Buffer Drain/Fills Minimum and maximum durations between jitter buffer drain or fill events, in seconds.

TranslatedCallingNumber, TranslatedCallingOctet,

TranslatedCalledNumber, TranslatedCalledOctet,

TranslatedRedirectCalled Number,

TranslatedRedirectCalledOctet

Translated call information.

TransmitBytes Number of bytes sent by this peer during this call.

TransmitPackets

TxDuration

Description

Total SIP call legs for which call records are available.

Total telephony call legs for which call records are available.

VAD

VoiceTxDuration

Number of packets sent by this peer during this call.

The length of the call. Appears only if the medium is

TELE.

Whether voice activation detection (VAD) was enabled for this call.

Duration of voice transmission from this peer to the voice gateway for this call. Derive the Voice

Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.

The following is sample output from the show call history fax briefcommand:

Router#

show call history fax brief

<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr> tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>

2 delay:<last>/<min>/<max>ms <codec>

Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm

: 5996450hs.25 +-1 +3802 pid:100 Answer 408 tx:0/0 rx:0/0 1F (T30 T1 EOM timeout)

Telephony : tx:38020/38020/0ms g729r8 noise:0dBm acom:0dBm

2 : 5996752hs.26 +-1 +3500 pid:110 Originate [email protected]

tx:0/0 rx:0/0 3F (The e-mail was not sent correctly. Remote SMTP server said: 354 )

3

IP 14.0.0.1 AcceptedMime:0 DiscardedMime:0

: 6447851hs.27 +1111 +3616 pid:310 Originate 576341.

tx:11/14419 rx:0/0 10 (Normal connection)

Telephony : tx:36160/11110/25050ms g729r8 noise:115dBm acom:-14dBm

3 : 6447780hs.28 +1182 +4516 pid:0 Answer tx:0/0 rx:0/0 10 (normal call clearing.)

IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0

4 : 6464816hs.29 +1050 +3555 pid:310 Originate 576341.

tx:11/14413 rx:0/0 10 (Normal connection)

4

Telephony : tx:35550/10500/25050ms g729r8 noise:115dBm acom:-14dBm

: 6464748hs.30 +1118 +4517 pid:0 Answer

5 tx:0/0 rx:0/0 10 (normal call clearing.)

IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0

: 6507900hs.31 +1158 +2392 pid:100 Answer 4085763413

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tx:0/0 rx:3/3224 10 (Normal connection)

Telephony : tx:23920/11580/12340ms g729r8 noise:0dBm acom:0dBm

5 : 6508152hs.32 +1727 +2140 pid:110 Originate [email protected]

tx:0/2754 rx:0/0 3F (service or option not available, unspecified)

IP 14.0.0.4 AcceptedMime:0 DiscardedMime:0

6 : 6517176hs.33 +1079 +3571 pid:310 Originate 576341.

tx:11/14447 rx:0/0 10 (Normal connection)

6

Telephony : tx:35710/10790/24920ms g729r8 noise:115dBm acom:-14dBm

: 6517106hs.34 +1149 +4517 pid:0 Answer

7 tx:0/0 rx:0/0 10 (normal call clearing.)

IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0

: 6567382hs.35 +1054 +3550 pid:310 Originate 576341.

tx:11/14411 rx:0/0 10 (Normal connection)

Telephony : tx:35500/10540/24960ms g729r8 noise:115dBm acom:-14dBm

7 : 6567308hs.36 +1128 +4517 pid:0 Answer tx:0/0 rx:0/0 10 (normal call clearing.)

IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0

The following example shows output for the show call history fax command with the T.38 Fax Relay statistics:

Router#

show call history fax

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 0

MGCP call-legs: 0

Total call-legs: 1

GENERIC:

SetupTime=9872460 ms

Index=8

PeerAddress=41023

PeerSubAddress=

PeerId=1

PeerIfIndex=242

LogicalIfIndex=180

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=9875610 ms

DisconnectTime=9936000 ms

CallDuration=00:01:00 sec

CallOrigin=2

ReleaseSource=1

ChargedUnits=0

InfoType=fax

TransmitPackets=268

TransmitBytes=4477

ReceivePackets=1650

ReceiveBytes=66882

TELE:

ConnectionId=[0xD6635DD5 0x9FA411D8 0x8005000A 0xF4107CA0]

IncomingConnectionId=[0xD6635DD5 0x9FA411D8 0x8005000A 0xF4107CA0]

CallID=7

Port=3/0/0:0 (7)

BearerChannel=3/0/0.8

TxDuration=6170 ms

VoiceTxDuration=0 ms

FaxTxDuration=0 ms

FaxRate=disable bps

FaxRelayMaxJitterBufDepth=560 ms

FaxRelayJitterBufOverFlow=0

FaxRelayMostRecentHSmodulation=V.17/short/14400

FaxRelayNumberOfPages=1

FaxRelayInitHSmodulation=V.17/long/14400

FaxRelayDirection=Transmit

FaxRelayPktLossConceal=0

FaxRelayEcmStatus=ENABLED

FaxRelayEncapProtocol=T.38 (UDPTL)

FaxRelayNsfCountryCode=Japan

FaxRelayNsfManufCode=0031B8EE80C48511DD0D0000DDDD0000DDDD000000000000000022ED00B0A400

FaxRelayFaxSuccess=Success

NoiseLevel=0

ACOMLevel=0

SessionTarget=

ImgPages=0

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CallerName=Analog 41023

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x80

OriginalCalledNumber=41021

OriginalCalledOctet=0xA1

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=41023

TranslatedCallingOctet=0x80

TranslatedCalledNumber=41021

TranslatedCalledOctet=0xA1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=41021

GwReceivedCalledOctet3=0xA1

The table below describes the fields not shown in the table above.

Table 35: show call history fax Field Descriptions

Field

FaxRelayDirection

FaxRelayEcmStatus

FaxRelayEncapProtocol

FaxRelayFaxSuccess

FaxRelayInitHSmodulation

FaxRelayMostRecentHSmodulation

FaxRelayNsfCountryCode

FaxRelayNsfManufCode

FaxRelayPktLossConceal

Description

Direction of fax relay.

Fax relay error correction mode status.

Fax relay encapsulation protocol.

Fax relay success.

Fax relay initial high speed modulation.

Fax relay most recent high speed modulation.

Fax relay Nonstandard Facilities (NSF) country code.

Fax relay NSF manufacturers code.

Fax relay packet loss conceal.

Related Commands

Command dial-control-mib show call active fax show call active voice show call history voice show dial-peer voice

Description

Specifies attributes for the call history table.

Displays call information for fax transmissions that are in progress.

Displays call information for voice calls that are in progress.

Displays the call history table for voice calls.

Displays configuration information for dial peers.

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show call history fax

Command show num-exp show voice port show call history fax through show debug condition

Description

Displays how the number expansions are configured in VoIP.

Displays configuration information about a specific voice port.

324

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show call history media

To display the call history table for media calls, use the show call history mediacommand in user EXEC or privileged EXEC mode.

show call history media [[brief] [id identifier]| compact [duration {less| more} seconds]| last number]

Syntax Description brief

id identifier

compact duration less more

seconds

last number

(Optional) Displays a truncated version of the call history table.

(Optional) Displays only the call with the specified

identifier. The range is from 1 to FFFF.

(Optional) Displays a compact version of the call history table.

(Optional) Displays the call history for the specified time duration.

Displays the call history for shorter duration calls.

Displays the call history for longer duration calls.

Time, in seconds. The range is from 1 to 2147483647.

(Optional) Displays the last calls connected, where the number of calls that appear is defined by the

numberargument. The range is from 1 to100 .

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

12.4(15)T

Modification

This command was introduced.

Usage Guidelines

This command displays a call-history table that contains a list of media calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number from 0 to 500 using the dial-control-mib command in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed, also specified by the dial-control-mib command. The default timer value is 15 minutes.

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Examples

You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the last keyword, and define the number of calls to be displayed with the number argument.

To display a truncated version of the call history table, use the brief keyword.

When a media call is active, you can display its statistics by using the show call active media command.

The following is sample output from the show call history mediacommand:

Router#

show call history media

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

Media call-legs: 4

Total call-legs: 4

GENERIC:

SetupTime=308530 ms

Index=4

PeerAddress=sip:[email protected]:5060

PeerSubAddress=

PeerId=2234

PeerIfIndex=184

LogicalIfIndex=0

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=309440 ms

DisconnectTime=320100 ms

CallDuration=00:00:10 sec

CallOrigin=1

ReleaseSource=7

ChargedUnits=0

InfoType=speech

TransmitPackets=237

TransmitBytes=37920

ReceivePackets=0

ReceiveBytes=0

VOIP:

ConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]

IncomingConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]

CallID=14

RemoteIPAddress=10.5.18.224

RemoteUDPPort=10002

RemoteSignallingIPAddress=10.5.18.224

RemoteSignallingPort=5060

RemoteMediaIPAddress=10.5.18.224

RemoteMediaPort=10002

SRTP = off

TextRelay = off

Fallback Icpif=0

Fallback Loss=0

Fallback Delay=0

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=rtp-nte

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2

[email protected]

SessionTarget=10.5.18.224

OnTimeRvPlayout=3000

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=2740 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=100 ms

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LoWaterPlayoutDelay=40 ms

Source tg label=test5

ReceiveDelay=90 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160 cvVoIPCallHistoryIcpif=16

MediaSetting=flow-around

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=555-0100

TranslatedCallingOctet=0x21

TranslatedCalledNumber=

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwOutpulsedCallingNumber=555-0101

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurationCallTimerStamp=

LongDurationCallDuration=

Username=

GENERIC:

SetupTime=308520 ms

Index=5

PeerAddress=sip:[email protected]:5060

PeerSubAddress=

PeerId=2235

PeerIfIndex=185

LogicalIfIndex=0

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=309370 ms

DisconnectTime=320100 ms

CallDuration=00:00:10 sec

CallOrigin=1

ReleaseSource=7

ChargedUnits=0

InfoType=speech

TransmitPackets=0

TransmitBytes=0

ReceivePackets=551

ReceiveBytes=88160

VOIP:

ConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]

IncomingConnectionId[0x2FB5B737 0xC3511DB 0x8005000B 0x5FDA0EF4]

CallID=13

RemoteIPAddress=10.5.18.224

RemoteUDPPort=10000

RemoteSignallingIPAddress=10.5.18.224

RemoteSignallingPort=5060

RemoteMediaIPAddress=10.5.18.224

RemoteMediaPort=10000

SRTP = off

TextRelay = off

Fallback Icpif=0

Fallback Loss=0

Fallback Delay=0

RoundTripDelay=0 ms

SelectedQoS=best-effort

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tx_DtmfRelay=rtp-nte

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2

[email protected]

SessionTarget=10.5.18.224

OnTimeRvPlayout=7000

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=2740 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=100 ms

LoWaterPlayoutDelay=40 ms

Source tg label=test5

ReceiveDelay=95 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160 cvVoIPCallHistoryIcpif=16

MediaSetting=flow-around

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=555-0102

TranslatedCallingOctet=0x21

TranslatedCalledNumber=

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwOutpulsedCallingNumber=555-0103

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurationCallTimerStamp=

LongDurationCallDuration=

Username=

GENERIC:

SetupTime=408050 ms

Index=7

PeerAddress=sip:[email protected]:5060

PeerSubAddress=

PeerId=2234

PeerIfIndex=184

LogicalIfIndex=0

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=408160 ms

DisconnectTime=426260 ms

CallDuration=00:00:18 sec

CallOrigin=1

ReleaseSource=7

ChargedUnits=0

InfoType=speech

TransmitPackets=598

TransmitBytes=95680

ReceivePackets=0

ReceiveBytes=0

VOIP:

ConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

IncomingConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

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CallID=19

RemoteIPAddress=10.5.18.224

RemoteUDPPort=10002

RemoteSignallingIPAddress=10.5.18.224

RemoteSignallingPort=5060

RemoteMediaIPAddress=10.5.18.224

RemoteMediaPort=10002

SRTP = off

TextRelay = off

Fallback Icpif=0

Fallback Loss=0

Fallback Delay=0

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=rtp-nte

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2

[email protected]

SessionTarget=10.5.18.224

OnTimeRvPlayout=11000

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=9560 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=100 ms

LoWaterPlayoutDelay=55 ms

Source tg label=test5

ReceiveDelay=100 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160 cvVoIPCallHistoryIcpif=16

MediaSetting=flow-around

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=555-0100

TranslatedCallingOctet=0x21

TranslatedCalledNumber=

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwOutpulsedCallingNumber=555-0101

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurationCallTimerStamp=

LongDurationCallDuration=

Username=

GENERIC:

SetupTime=408040 ms

Index=8

PeerAddress=sip:[email protected]:5060

PeerSubAddress=

PeerId=2235

PeerIfIndex=185

LogicalIfIndex=0

DisconnectCause=10

DisconnectText=normal call clearing (16)

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ConnectTime=408130 ms

DisconnectTime=426260 ms

CallDuration=00:00:18 sec

CallOrigin=1

ReleaseSource=7

ChargedUnits=0

InfoType=speech

TransmitPackets=0

TransmitBytes=0

ReceivePackets=911

ReceiveBytes=145760

VOIP:

ConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

IncomingConnectionId[0x6B02FC0C 0xC3511DB 0x8006000B 0x5FDA0EF4]

CallID=18

RemoteIPAddress=10.5.18.224

RemoteUDPPort=10000

RemoteSignallingIPAddress=10.5.18.224

RemoteSignallingPort=5060

RemoteMediaIPAddress=10.5.18.224

RemoteMediaPort=10000

SRTP = off

TextRelay = off

Fallback Icpif=0

Fallback Loss=0

Fallback Delay=0

RoundTripDelay=0 ms

SelectedQoS=best-effort tx_DtmfRelay=rtp-nte

FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2

[email protected]

SessionTarget=10.5.18.224

OnTimeRvPlayout=9000

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=9560 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=100 ms

LoWaterPlayoutDelay=55 ms

Source tg label=test5

ReceiveDelay=100 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

VAD = disabled

CoderTypeRate=g711ulaw

CodecBytes=160 cvVoIPCallHistoryIcpif=16

MediaSetting=flow-around

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=

OriginalCallingOctet=0x0

OriginalCalledNumber=

OriginalCalledOctet=0x0

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x0

TranslatedCallingNumber=555-0100

TranslatedCallingOctet=0x21

TranslatedCalledNumber=

TranslatedCalledOctet=0xC1

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwOutpulsedCallingNumber=555-0101

GwOutpulsedCallingOctet3=0x21

GwOutpulsedCallingOctet3a=0x81

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

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LongDurationCallDetected=no

LongDurationCallTimerStamp=

LongDurationCallDuration=

Username=

The table below describes the significant fields shown in the display, in alphabetical order.

Table 36: show call history media Field Descriptions

Field

CallDuration

CallOrigin

ChargedUnits

CodecBytes

CoderTypeRate

ConnectionId

ConnectTime

GapFillWithInterpolation

GapFillWithRedundancy

GapFillWithPrediction

GapFillWithSilence

Description

Length of the call, in hours, minutes, and seconds, hh:mm:ss.

Call origin: not answer or originate.

Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.

Payload size, in bytes, for the codec used.

Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.

Global call identifier for this gateway call.

Time, in ms, during which the call was connected.

Duration, in ms, of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

Duration, in ms, of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.

Duration, in ms, of the voice signal played out with a signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and

G.723.1 compression algorithms.

Duration, in ms, of a voice signal replaced with silence because voice data was lost or not received in time for this call.

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show call history media

PeerAddress

PeerId

PeerIfIndex

PeerSubAddress

ReceiveBytes

ReceiveDelay

ReceivePackets

ReleaseSource

RemoteIPAddress

Field

GENERIC

H323 call-legs

HiWaterPlayoutDelay

Index

InfoType

LogicalIfIndex

LoWaterPlayoutDelay

OnTimeRvPlayout

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Description

Generic or common parameters; that is, parameters that are common for VoIP and telephony call legs.

Total H.323 call legs for which call records are available.

High-water-mark voice playout first in first out

(FIFO) Delay during this call, in ms.

Dial peer identification number.

Information type for this call; for example, voice, speech, or fax.

Index number of the logical interface for this call.

Low-water-mark voice playout FIFO delay during this call, in ms.

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Destination pattern or number associated with this peer.

ID value of the peer table entry to which this call was made.

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

Subaddress when this call is connected.

Number of bytes received by the peer during this call.

Average playout FIFO delay plus the decoder delay during this voice call, in ms.

Number of packets received by this peer during this call.

Number value of the release source.

Remote system IP address for the VoIP call.

show call history fax through show debug condition

Field

RemoteUDPPort

RoundTripDelay

SelectedQoS

SessionProtocol

SessionTarget

SetupTime

SIP call-legs

Telephony call-legs

TransmitBytes

TransmitPackets

VAD

Related Commands

Command dial-control-mib show call active media show call history media

Description

Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.

Voice packet round-trip delay, in ms, between the local and remote systems on the IP backbone for this call.

Selected Resource Reservation Protocol (RSVP) quality of service (QoS) for this call.

Session protocol used for an Internet call between the local and remote routers through the IP backbone.

Session target of the peer used for this call.

Value of the system UpTime, in ms, when the call associated with this entry was started.

Total Session Initiation Protocol (SIP) call legs for which call records are available.

Total telephony call legs for which call records are available.

Number of bytes sent by this peer during this call.

Number of packets sent by this peer during this call.

Whether voice activation detection (VAD) was enabled for this call.

Description

Sets the maximum number of calls contained in the table.

Displays call information for media calls in progress.

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show call history fax through show debug condition show call history video

show call history video

To display call history information for signaling connection control protocol (SCCP) video calls, use the show

call history video command in user EXEC or privileged EXEC mode.

show call history video [[brief] [id identifier]| compact [duration {less| more} seconds]| last number]

Syntax Description brief

id identifier

compact duration less more

seconds

last number

(Optional) Displays a truncated version of video call history information.

(Optional) Displays only the video call history with the specified identifier. Range is a hexadecimal value from 1 to FFFF.

(Optional) Displays a compact version of video call history information.

(Optional) Displays the call history for the specified time duration.

Displays the call history for shorter duration calls.

Displays the call history for longer duration calls.

Time, in seconds. The range is from 1 to 2147483647.

(Optional) Displays the last calls connected, where the number of calls that appear is defined by the

numberargument. The range is from 1 to100 .

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Cisco IOS Release

12.4(4)XC

12.4(9)T

Cisco Product

Cisco Unified CME 4.0

Cisco Unified CME 4.0

12.4(16); 12.4(15)T Cisco Unified CME 4.0

Modification

This command was introduced.

This command was integrated into Cisco IOS

Release 12.4(9)T.

This command was modified. The Port and

BearerChannel display fields were added to the

TELE call leg record of the command output.

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Examples

The following is sample output from the show call history video command with the compact option:

Router#

show call history video compact

<callID> A/O FAX T<sec> Codec

Total call-legs: 2

241

242

ANS

ORG

T17

T17 g729r8 g729r8 type Peer Address

VOIP P555-0100

TELE-VIDEO P555-0101

The table below describes the significant fields shown in the display.

Table 37: show call history video Field Descriptions

IP R<ip>:<udp>

192.0.2.0:16926

Field

callID

A/O

FAX

T<sec>

Codec type

Peer Address

IP R<ip>:<udp>

Total call-legs

Description

Unique identifier for the call leg.

Call leg was an answer (ANS) or an originator (ORG).

Fax number for the call leg.

Duration in seconds.

Codec used for this call leg.

Call type for this call leg.

Called or calling number of the remote peer.

IP address and port number

Total number of call legs for this call.

Related Commands

Command show call active video

Description

Displays call information for SCCP video calls in progress.

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show call history video record

To display information about incoming and outgoing video calls, use the s how call history video

recordcommand in privileged EXEC mode.

show call history video record

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)XK

12.0(7)T

Modification

This command was introduced on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.0(7)T.

Examples

The following example displays information about two video calls:

Router#

show call history video record

CallId = 4

CalledNumber = 221

CallDuration = 39006 seconds

DisconnectText = remote hangup

SVC: call ID = 8598630

Remote NSAP = 47.0091810000000002F26D4901.00107B09C645.C8

Local NSAP = 47.0091810000000002F26D4901.00107B4832E1.C8

vcd = 414, vpi = 0, vci = 158

SerialPort = Serial0

VideoSlot = 1, VideoPort = 0

CallId = 3

CalledNumber = 221

CallDuration = 557 seconds

DisconnectText = local hangup

SVC: call ID = 8598581

Remote NSAP = 47.0091810000000002F26D4901.00107B09C645.C8

Local NSAP = 47.0091810000000002F26D4901.00107B4832E1.C8

vcd = 364, vpi = 0, vci = 108

SerialPort = Serial0

VideoSlot = 1, VideoPort = 0

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show call history voice

To display the call history table for voice calls, use the show call history voice command in user EXEC or privileged EXEC mode.

show call history voice [brief [id identifier]| compact [duration {less| more} seconds]| dest-route-string

tag| id identifier| last number| redirect {rtpvt| tbct}| stats]

Syntax Description brief

id identifier

compact

dest-route-string tag

duration seconds

last number

redirect

(Optional) Displays a truncated version of the call history table.

(Optional) Displays only the call with the specified identifier. Range is from 1 to FFFF.

(Optional) Displays a compact version of the call history table.

(Optional) Displays only the call with the specified destination route tag value. The range is from 1 to

10000.

(Optional) Displays history information for calls that are longer or shorter than the value of the specified

seconds argument. The arguments and keywords are as follows:

less --Displays calls shorter than the seconds value.

more --Displays calls longer than the seconds value.

seconds --Elapsed time, in seconds. Range is from 1 to 2147483647.

(Optional) Displays the last calls connected, where the number of calls that appear is defined by the

numberargument. Range is from 1 to100 .

(Optional) Displays information about calls that were redirected using Release-to-Pivot (RTPvt) or Two

B-Channel Transfer (TBCT). The keywords are as follows:

rtpvt --Displays information about RTPvt calls.

tbct --Displays information about TBCT calls.

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show call history fax through show debug condition show call history voice stats

(Optional) Displays information about digital signal processing (DSP) voice quality metrics.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

11.3(1)T

12.0(3)XG

12.0(4)XJ

12.0(4)T

12.0(5)XK

12.0(7)XK

12.0(7)T

12.1(2)T

12.1(5)XM

12.1(5)XM2

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(8)T

12.2(11)T

12.2(13)T

12.3(1)

Modification

This command was introduced on the Cisco 3600 series.

Support was added for Voice over Frame Relay (VoFR) on the Cisco 2600 series and Cisco 3600 series.

This command was modified for store-and-forward fax.

The brief keyword was added, and the command was implemented on the

Cisco 7200 series.

This command was implemented on the Cisco MC3810.

The brief keyword was implemented on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.0(7)T.

This command was integrated into Cisco IOS Release 12.1(2)T.

This command was implemented on the Cisco AS5800.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

The output of this command was modified to indicate whether a specified call has been established using Annex E.

This command was integrated into Cisco IOS Release 12.2(4)T.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(8)T. Support was not included for the Cisco AS5350, Cisco AS5400, Cisco AS5800, and

Cisco AS5850.

Support was added for Cisco AS5350, Cisco AS5400, Cisco AS5800, and

Cisco AS5850.

The ReleaseSource field was added to the Field Description table, and the

record keyword was deleted from the command name.

The redirect keyword was added.

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Release

12.4(2)T

12.4(11)XW

12.4(15)T

12.4(16)

Modification

The LocalHostname display field was added to the VoIP call leg record.

The stats keyword was added.

The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

The Port and BearerChannel display fields were added to the TELE call leg record of the command output.

12.4(22)T

15.3(3)M

Command output was updated to show IPv6 information.

This command was modified. The dest-route-string keyword was added.

Cisco IOS XE Release 3.10S

This command was integrated into Cisco IOS XE Release 3.10S.

Usage Guidelines

This command displays a call-history table that contains a list of voice calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number from 0 to 500 using the dial-control-mib command in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed.

The timer value is also specified by the dial-control-mib command. The default timer value is 15 minutes.

You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the last keyword, and define the number of calls to be displayed with the number argument.

To display a truncated version of the call history table, use the brief keyword.

Use the show call active voice redirect command to review records for calls that implemented RTPvt or

TBCT.

When a call is active, you can display its statistics by using the show call active voice command.

Use the show call active voice dest-route-string command to display only the active voice calls with call routing configured using specified destination-route-string globally and at the dial-peer level.

Examples

The following is sample output from the show call history voice command:

Router#

show call history voice

GENERIC:

SetupTime=104648 ms

Index=1

PeerAddress=55240

PeerSubAddress=

PeerId=2

PeerIfIndex=105

LogicalIfIndex=0

DisconnectCause=10

DisconnectText=normal call clearing.

ConnectTime=104964

DisconectTime=143329

CallDuration=00:06:23

CallOrigin=1

ChargedUnits=0

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show call history voice

InfoType=speech

TransmitPackets=37668

TransmitBytes=6157536

ReceivePackets=37717

ReceiveBytes=6158452

VOIP:

ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]

CallID=2

RemoteIPAddress=10.14.82.14

RemoteUDPPort=18202

RoundTripDelay=2 ms

SelectedQoS=best-effort tx_DtmfRelay=inband-voice

FastConnect=TRUE

SessionProtocol=cisco

SessionTarget=ipv4:10.14.82.14

OnTimeRvPlayout=40

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=67 ms

LoWaterPlayoutDelay=67 ms

ReceiveDelay=67 ms

LostPackets=0 ms

EarlyPackets=0 ms

LatePackets=0 ms

VAD = enabled

CoderTypeRate=g729r8

CodecBytes=20 cvVoIPCallHistoryIcpif=0

SignalingType=cas

Modem passthrough signaling method is nse

Buffer Fill Events = 0

Buffer Drain Events = 0

Percent Packet Loss = 0

Consecutive-packets-lost Events = 0

Corrected packet-loss Events = 0

Last Buffer Drain/Fill Event = 373sec

Time between Buffer Drain/Fills = Min 0sec Max 0sec

GENERIC:

SetupTime=104443 ms

Index=2

PeerAddress=50110

PeerSubAddress=

PeerId=100

PeerIfIndex=104

LogicalIfIndex=10

DisconnectCause=10

DisconnectText=normal call clearing.

ConnectTime=104964

DisconectTime=143330

CallDuration=00:06:23

CallOrigin=2

ChargedUnits=0

InfoType=speech

TransmitPackets=37717

TransmitBytes=5706436

ReceivePackets=37668

ReceiveBytes=6609552

TELE:

ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]

CallID=3

Port=3/0/0 (3)

BearerChannel=3/0/0.1

TxDuration=375300 ms

VoiceTxDuration=375300 ms

FaxTxDuration=0 ms

CoderTypeRate=g711ulaw

NoiseLevel=-75

ACOMLevel=11

SessionTarget=

ImgPages=0

show call history fax through show debug condition

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The following example from a Cisco AS5350 router displays a sample of voice call history records showing release source information:

Router#

show call history voice

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 1

Total call-legs: 2

GENERIC:

SetupTime=85975291 ms

.

.

.

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=85975335

DisconnectTime=85979339

CallDuration=00:00:40

CallOrigin=1

ReleaseSource=1

.

.

.

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=85975335

DisconnectTime=85979339

CallDuration=00:00:40

CallOrigin=1

ReleaseSource=1

.

.

.

.

.

.

VOIP:

ConnectionId[0x2868AD84 0x375B11D4 0x8012F7A5 0x74DE971E]

CallID=1

GENERIC:

SetupTime=85975290 ms

.

.

.

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=85975336

DisconnectTime=85979340

CallDuration=00:00:40

CallOrigin=2

ReleaseSource=1

.

.

.

TELE:

ConnectionId=[0x2868AD84 0x375B11D4 0x8012F7A5 0x74DE971E]

CallID=2

Port=3/0/0 (2)

BearerChannel=3/0/0.1

The following is sample output from the show call history voice brief command:

Router#

show call history voice brief

<ID>: <CallID> <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr> dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)

IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late> delay:<last>/<min>/<max>ms <codec> media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected> last <buf event time>s dur:<Min>/<Max>s

FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

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<codec> (payload size)

ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

<codec> (payload size)

Telephony <int> (callID) [channel_id] tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm

MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> disc:<cause code> speeds(bps): local <rx>/<tx> remote <rx>/<tx>

Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf> bw: <req>/<act> codec: <audio>/<video> tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes> rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

The following is sample output from the show call history voice redirect command:

Router#

show call history voice redirect tbct

index=2, xfr=tbct-notify, status=redirect_success, start_time=*00:12:25.981 UTC Mon Mar 1

1993, ctrl name=T1-2/0, tag=13 index=3, xfr=tbct-notify, status=redirect_success, start_time=*00:12:25.981 UTC Mon Mar 1

1993, ctrl name=T1-2/0, tag=13 index=4, xfr=tbct-notify, status=redirect_success, start_time=*00:13:07.091 UTC Mon Mar 1

1993, ctrl name=T1-2/0, tag=12 index=5, xfr=tbct-notify, status=redirect_success, start_time=*00:13:07.091 UTC Mon Mar 1

1993, ctrl name=T1-2/0, tag=12

Number of call-legs redirected using tbct with notify:4

The table below describes the significant fields shown in the show call history voice redirect tbct display.

Table 38: show call history voice redirect Field Descriptions

Field

index xfr

Description

Index number of the record in the history file.

Whether TBCT or TBCT with notify has been invoked.

status start_time ctrl name

Status of the redirect request.

Time, in hours, minutes, and seconds when the redirected call began.

Name of the T1 controller where the call originated.

tag Call tag number that identifies the call.

Number of call-legs redirected using tbct with notify Total number of call legs that were redirected using

TBCT with notify.

Related Commands

Command dial-control-mib show call active fax

Description

Set the maximum number of calls contained in the table.

Displays call information for fax transmissions that are in progress.

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Command show call active voice show call history fax show dial-peer voice show num-exp show voice port show call history voice

Description

Displays call information for voice calls that are in progress.

Displays the call history table for fax transmissions.

Displays configuration information for dial peers.

Displays how the number expansions are configured in VoIP.

Displays configuration information about a specific voice port.

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show call history fax through show debug condition show call language voice

show call language voice

To display a summary of languages configured and the URLs of the corresponding Tool Command Language

(TCL) modules for the languages that are not built-in languages, use the show call language voice command

inEXEC mode.

show call language voice [language| summary]

Syntax Description

language

summary

(Optional) Two-character prefix configured with the

call language voice command in global configuration mode, either for a prefix for a built-in language or one that you have defined; for example, "en" for

English or "ru" for Russian.

(Optional) Summary of all the languages configured and the URLs for the TCL modules other than built-in languages.

Command Modes

EXEC (#)

Command History

Release

12.2(2)T

Modification

This command was introduced.

Usage Guidelines

This command is similar to the show call application voice command. If a language is built in, the URL listed reads "fixed." If you decide to overwrite the built-in language with your own language, the word "fixed" in the URL column changes to the actual URL where your new application lives.

Examples

The following command displays a summary of the configured languages:

Router# show call language voice summary name url sp ch fixed fixed en ru fixed tftp://dirt/fwarlau/scripts/multilag/ru_translate.tcl

The following command displays information about Russian-language configuration:

Router# show call language voice ru ru_translate.tcl

ru_translate.tcl~ singapore.cfg

test.tcl

people% more ru_translate.tcl

# Script Locked by: farmerj

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# Script Version: 1.1.0.0

# Script Lock Date: Sept 24 2000

# ca_translate.tcl

#------------------------------------------------------------------

# Sept 24, 2000 Farmer Joe

#

# Copyright (c) 2000 by Cisco Systems, Inc.

# All rights reserved.

#------------------------------------------------------------------

#<snip>...

...set prefix ""

#puts "argc"

#foreach arg $argv {

#puts "$arg"

#

#

#} translates $arg puts "\t\t**** $prompt RETURNED"

Field descriptions should be self-explanatory.

Related Commands

Command call language voice call language voice load debug voip ivr show call application voice

Description

Configures a TCL module.

Loads or reloads a TCL module from the configured

URL location.

Specifies the type of VoIP IVR debug output that you want to view.

Shows and describes applications.

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show call history fax through show debug condition show call leg

show call leg

To display event logs and statistics for voice call legs, use the show call legcommand in privileged EXEC mode.

show call leg {active| history} [summary| [last number| leg-id leg-id] [event-log| info]]

Syntax Description active history

summary

last number

leg-id leg-id event-log info

Statistics or event logs for active call legs.

Statistics or event logs for terminated call legs.

(Optional) A summary of each call leg.

(Optional) Selected number of most recent call legs.

Not available with active keyword.

(Optional) A specific call leg. Output displays event logs or statistics for that call leg.

(Optional) Event logs for call legs.

(Optional) Statistics for call legs.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(8)T

Modification

This command was introduced.

Usage Guidelines

If you use the leg-id keyword, only statistics or event logs for that call leg display. To display event logs with this command, you must enable event logging with the call leg event-log command.

Examples

The following is sample output from the show call leg command using different keywords:

Router#

show call leg active summary

G<id> L<id> Elog A/O FAX T<sec> Codec

G11DC L A

Total call-legs: 1

Y ANS T2 None

Router#

show call leg active event-log

type

TELE

Peer Address

P4085550198

IP R<ip>:<udp>

Event log for call leg ID: A buf_size=4K, log_lvl=INFO

Connection ID: 11DC

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927,

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echo canceller = enable, direct inward dialing

A:1057277701:72:INFO: Dialpeer = 1

A:1057277701:77:INFO: Digit collection

A:1057277701:78:INFO: Call connected using codec None

Total call-legs: 1

Router#

show call leg active info

Information for call leg ID: A

GENERIC:

SetupTime=3012940 ms

Index=1

PeerAddress=4085550198

PeerSubAddress=

PeerId=1

PeerIfIndex=329

LogicalIfIndex=253

ConnectTime=301295

CallDuration=00:00:20

CallState=4

CallOrigin=2

ChargedUnits=0

Connection ID: 11DC

InfoType=2

TransmitPackets=412

TransmitBytes=98880

ReceivePackets=0

ReceiveBytes=0

TELE:

ConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]

IncomingConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]

TxDuration=20685 ms

VoiceTxDuration=0 ms

FaxTxDuration=0 ms

CoderTypeRate=None

NoiseLevel=-120

ACOMLevel=90

OutSignalLevel=-50

InSignalLevel=-41

InfoActivity=0

ERLLevel=38

EchoCancellerMaxReflector=16685

SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=4085550198

OriginalCallingOctet=0x0

OriginalCalledNumber=52927

OriginalCalledOctet=0xE9

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=4085550198

TranslatedCallingOctet=0x0

TranslatedCalledNumber=52927

TranslatedCalledOctet=0xE9

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=52927

GwReceivedCalledOctet3=0xE9

GwReceivedCallingNumber=4085550198

GwReceivedCallingOctet3=0x0

GwReceivedCallingOctet3a=0x81

Total call-legs: 1

For a description of the call leg statistics, see the description for the show call active voice command.

Router#

show call leg active leg-id A

Call Information - Connection ID: 11DC , Call Leg ID: A

GENERIC:

SetupTime=3012940 ms

Index=1

PeerAddress=4085550198

PeerSubAddress=

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show call history fax through show debug condition show call leg

PeerId=1

PeerIfIndex=329

LogicalIfIndex=253

ConnectTime=301295

CallDuration=00:00:40

CallState=4

CallOrigin=2

ChargedUnits=0

InfoType=2

TransmitPackets=824

TransmitBytes=197760

ReceivePackets=0

ReceiveBytes=0

TELE:

ConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]

IncomingConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]

TxDuration=20685 ms

VoiceTxDuration=0 ms

FaxTxDuration=0 ms

CoderTypeRate=None

NoiseLevel=-120

ACOMLevel=90

OutSignalLevel=-50

InSignalLevel=-41

InfoActivity=0

ERLLevel=38

EchoCancellerMaxReflector=16685

SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=4085550198

OriginalCallingOctet=0x0

OriginalCalledNumber=52927

OriginalCalledOctet=0xE9

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=4085550198

TranslatedCallingOctet=0x0

TranslatedCalledNumber=52927

TranslatedCalledOctet=0xE9

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=52927

GwReceivedCalledOctet3=0xE9

GwReceivedCallingNumber=4085550198

GwReceivedCallingOctet3=0x0

GwReceivedCallingOctet3a=0x81

Call Event Log - Connection ID: 11DC , Call Leg ID: A buf_size=4K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing

A:1057277701:72:INFO: Dialpeer = 1

A:1057277701:77:INFO: Digit collection

A:1057277701:78:INFO: Call connected using codec None

Call-leg found: 1

Router#

show call leg active leg-id A event-log

Call Event Log - Connection ID: 11DC , Call Leg ID: A buf_size=4K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing

A:1057277701:72:INFO: Dialpeer = 1

A:1057277701:77:INFO: Digit collection

A:1057277701:78:INFO: Call connected using codec None

Call-leg found: 1

Router#

show call leg history summary

G<id> L<id> disc-cause

G11DB L 7

Elog A/O FAX T<sec> Codec

Y ANS T24 None type Peer Address

TELE P4085550198 D10

IP R<ip>:<udp>

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G11DC L A

Total call-legs: 2

Y ANS T159 None TELE

Router#

show call leg history last 1

Call Information - Connection ID: 11DC , Call Leg ID: A

GENERIC:

P4085550198 D10

SetupTime=3012940 ms

Index=4

PeerAddress=4085550198

PeerSubAddress=

PeerId=1

PeerIfIndex=329

LogicalIfIndex=253

DisconnectCause=10

DisconnectText=normal call clearing (16)

ConnectTime=301295

DisconnectTime=317235

CallDuration=00:02:39

CallOrigin=2

ReleaseSource=1

ChargedUnits=0

InfoType=speech

TransmitPackets=2940

TransmitBytes=705600

ReceivePackets=0

ReceiveBytes=0

TELE:

ConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]

IncomingConnectionId=[0x632D2CAB 0xACEB11D7 0x80050030 0x96F8006E]

TxDuration=20685 ms

VoiceTxDuration=0 ms

FaxTxDuration=0 ms

CoderTypeRate=None

NoiseLevel=-120

ACOMLevel=90

SessionTarget=

ImgPages=0

CallerName=

CallerIDBlocked=False

OriginalCallingNumber=4085550198

OriginalCallingOctet=0x0

OriginalCalledNumber=52927

OriginalCalledOctet=0xE9

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0xFF

TranslatedCallingNumber=4085550198

TranslatedCallingOctet=0x0

TranslatedCalledNumber=52927

TranslatedCalledOctet=0xE9

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0xFF

GwReceivedCalledNumber=52927

GwReceivedCalledOctet3=0xE9

GwReceivedCallingNumber=4085550198

GwReceivedCallingOctet3=0x0

GwReceivedCallingOctet3a=0x81

Call Event Log - Connection ID: 11DC , Call Leg ID: A buf_size=4K, log_lvl=INFO

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing

A:1057277701:72:INFO: Dialpeer = 1

A:1057277701:77:INFO: Digit collection

A:1057277701:78:INFO: Call connected using codec None

A:1057277860:150:INFO: Inform application call disconnected (cause = normal call clearing

(16))

A:1057277860:154:INFO: Call disconnected (cause = normal call clearing (16))

A:1057277860:155:INFO: Call released

Total call-legs: 1

Total call-legs with event log: 1

Router#

show call leg history leg-id A event-log

Call Event Log - Connection ID: 11DC , Call Leg ID: A buf_size=4K, log_lvl=INFO

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show call history fax through show debug condition show call leg

<ctx_id>:<timestamp>:<seq_no>:<severity>:<msg_body>

A:1057277701:71:INFO: Call setup indication received, called = 4085550198, calling = 52927, echo canceller = enable, direct inward dialing

A:1057277701:72:INFO: Dialpeer = 1

A:1057277701:77:INFO: Digit collection

A:1057277701:78:INFO: Call connected using codec None

A:1057277860:150:INFO: Inform application call disconnected (cause = normal call clearing

(16))

A:1057277860:154:INFO: Call disconnected (cause = normal call clearing (16))

A:1057277860:155:INFO: Call released

Call-leg matched ID found: 1

Call-legs matched ID with event log: 1

Field descriptions should be self-explanatory.

Related Commands

Command call leg event-log call leg event-log dump ftp call leg event-log error-only call leg event-log max-buffer-size call leg history event-log save-exception-only monitor call leg event-log

Description

Enables event logging for voice, fax, and modem call legs.

Enables the voice gateway to write the contents of the call-leg event log buffer to an external file.

Restricts event logging to error events only for voice call legs.

Sets the maximum size of the event log buffer for each call leg.

Saves to history only event logs for call legs that had at least one error.

Displays the event log for an active call leg in real-time.

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show call history fax through show debug condition show call media forking

show call media forking

To display currently active media forking sessions, use the show call media forking command in user EXEC or privileged EXEC mode.

show call media forking

Syntax Description

This command has no arguments or keywords.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

15.2(2)T

Modification

This command was introduced.

Usage Guidelines

Use this command to verify that media forking was successful for relevant anchor legs.

Examples

The following example is a sample output from the show call media forking command..

Router#

show call media forking

Warning: Output may be truncated if sessions are added/removed concurrently!

Session Call n/f Destination (port address)

7 6 far 1234 1.5.35.254

8 6 near 5678 1.5.35.254

The table below describes the fields that are displayed in the output.

Field

Session

Description

Session Identifier.

Call n/f

Call Leg identifier in hexadecimal. It must match the

Call ID from the show call leg active command.

Direction (Near End or Far End) of the voice stream that was forked.

Destination (port address) Destination for the forked packets. It consists of the folllowing:

• RTP Port

• IP Address

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show call history fax through show debug condition show callmon

show callmon

To display call monitor information, use the show callmon command in user EXEC or privileged EXEC mode.

show callmon {call| gcid| subscription| trace {all| event {all| call| connection}| exec| server| subscription|

trigger}}

Syntax Description call gcid subscription trace all event exec server subscription trigger

Displays the active call monitor calls.

Displays the active global call ID information.

Displays the subscription information.

Displays the trace information.

Displays all types of traces based on time.

Displays the event trace information.

all --Displays all event traces.

call --Displays event traces related to a call.

connection --Displays the event traces related to a connection.

Displays all critical execution traces.

Displays all session server up or down traces.

Displays all subscription traces.

Displays the entire trigger structure by index.

Command Modes

User EXEC (>) Privileged EXEC (#))

Command History

Release

12.4(22)T

Modification

This command was introduced.

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Examples

The following sample output from the show callmon call command shows active call monitor calls:

Router#

show callmon call

line dn sub_id number of call instance

6401,

6601,

1 callID 2038(19D7), *cg = 6401, cd = 6601

1 callID 2039(19D7), cg = 6401, *cd = 6601

The table below describes the significant fields shown in the display.

Table 39: show callmon call Field Descriptions

Field

dn number of call instance

Description

Directory number.

Number of call instances.

Contents of the call instance.

The following sample output from the show callmon gcid command shows the active global call ID information:

Router#

show callmon gcid

GCID callIDs(active_entry_id)

AE48ECBC-D89311DB-87FC996E-115FF692 isConfGcid:FALSE gcid_conf:00000000-00000000-00000000-00000000

, 2038(19D7), 2039(19D7)

The table below describes the significant fields shown in the display.

Table 40: show callmon gcid Field Descriptions

Field

GCID

CallIDs

Description

Global call ID.

Active call IDs.

Related Commands

Command callmonitor

Description

Enables call monitoring messaging functionality on a SIP endpoint in a VoIP network.

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show call history fax through show debug condition show call prompt-mem-usage

show call prompt-mem-usage

To display the amount of memory used by prompts, use the show call prompt-mem-usagecommand in privileged EXEC mode.

show call prompt-mem-usage [detail]

Syntax Description detail

(Optional) Displays details about memory usage and names of tones used.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(15)T

12.3(7)T

Modification

This command was introduced.

The detail keyword was added.

Usage Guidelines

Use this command to display the number of prompts loaded into the gateway, the amount of memory used by the prompts, the number of prompts currently being played, and the status of prompt loads.

For calls transferred by a Cisco CallManager Express (Cisco CME) system, the ringback tone generation for commit-at-alerting uses an interactive voice response (IVR) prompt playback mechanism. Ringback tone is played to the transferred party by the Cisco CME system associated with the transferring party.

The system automatically generates tone prompts as needed on the basis of the network-locale setting made in the Cisco CME system.

Examples

The following sample output shows details about the memory usage of the prompts that are used.

Router#

show call prompt-mem-usage

Prompt memory usage: file(s) config

’d

0200 wait

0010 active

0001 free

00189 memory 02097152 00081259 00055536 01960357

Prompt load counts: (counters reset 0) success 11(1st try) 0(2nd try), failure 0

Other mem block usage: mcDynamic mcReader gauge 00001 00001

Number of prompts playing: 1 mc total

00011

00136795

Number of start delays : 0

MCs in the ivr MC sharing table

===============================

Media Content: NoPrompt (0x83C64554)

URL: cid=0, status=MC_READY size=24184 coding=g711ulaw refCount=0 ms total

00002

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Media Content: tone://GB_g729_tone_ringback (0x83266EC8)

URL: tone://GB_g729_tone_ringback

The table below describes the significant fields shown in the display.

Table 41: show call prompt-mem-usage Field Descriptions

Field

file(s) file(s) - config’d file(s) -wait file(s) - active file(s) - free file(s) - mc total ms total memory memory - config’d memory - wait memory - active memory - free

Description

Number of prompts in different queues.

Maximum number of configured prompts that can be simultaneously available in memory. In the sample output, the value of 200 in this field means that loading the 201st prompt results in the oldest prompts being removed.

Number of prompts in the wait queue that are not being used in any call and are ready to be deleted when there is no space for a new prompt. This field lists older prompts that can be deleted.

Number of prompts that are being used in active calls.

These prompts cannot be deleted.

Number of prompts that can be loaded without deleting any prompt from the wait queue. This is the number of configured prompts (listed under config’d) minus the total number of prompts in the wait and active states.

Total number of prompts in the wait and active states.

Number of media streams that are currently active.

One media stream is used for playing INBOX prompts. A prompt is considered an INBOX prompt if its URL is either flash:, http:, ram:, or tftp:.

Displays the memory used by prompts, in bytes.

Maximum amount of memory configured to be available for prompts.

Total amount of memory used by prompts in the wait list.

Total amount of memory used by prompts in the active list.

Amount of available memory. This is the amount of configured prompts (listed under config’d) memory minus the total amount of memory used by the prompts in the wait and active lists.

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show call prompt-mem-usage

Field

memory - mc total

Prompt load counts mcDynamic mcReader

Number of prompts playing

Number of start delays

MCs in the ivr MC sharing table

Media Content

URL cid

show call history fax through show debug condition

Description

Total amount of memory used by prompts in the wait and active lists.

Number of successful attempts to load a prompt on the first try and on the second try, and the number of attempts to load a prompt that failed.

Number of dynamic element queues that are active.

A dynamic element queue is a list of prompts that are played together.

Number of mcReaders that are active. An mcReader is used for playing one mcDynamic queue of prompts.

An mcReader is used only if the mcDynamic contains prompts that are associated with one of the following types of URL: flash:, http:, ram:, or tftp:.

Number of prompts that are currently playing.

Number of times that prompts failed to start and have subsequently restarted.

The fields below this line of text refer to each media content (prompt) currently cached in memory. In the sample output, the only cached prompt is the built-in default prompt named "NoPrompt."

Name of the prompt, which is derived from the audio file URL (the characters after the last "/" in the URL).

The address in parentheses is the memory location of the prompt.

Location of the file for the prompt that is playing. In the case of the default prompt, NoPrompt, no URL is given.

Call identification number of the call that initiated the loading of the prompt.

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Field

status size coding refCount=0

show call prompt-mem-usage

Description

Status of the media content. The following values are possible:

• MC_NOT_READY--Initial status for media content. When the media content is successfully loaded, the status will change to MC_READY.

• MC_READY--Media content is loaded into memory and ready for use.

• MC_LOAD_FAIL--Media content failed to load.

Size of the media content, in bytes.

Type of encoding used by the media content.

Number of calls to which this media content is currently being streamed.

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show call history fax through show debug condition show call resource voice stats

show call resource voice stats

To display resource statistics for an H.323 gateway, use the show call resource voice stats command in privileged EXEC mode.

show call resource voice stats [ds0| dsp]

Syntax Description ds0 dsp

(Optional) Specifies the voice digital signal level zero

(DS0) resource statistics information.

(Optional) Specifies the voice digital signal processor

(DSP) resource statistics information.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)T

12.1(5)XM2

12.2(4)T

12.2(2)XB1

12.2(8)T

12.4(22)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.1(5)XM2

This command was integrated into Cisco IOS Release 12.2(4)T.

This command was integrated into Cisco IOS Release into 12.2(2)XB1.

This command was modified. Support for the Cisco AS5300,Cisco AS5350,

Cisco AS5400, Cisco AS5800, and Cisco AS5850 series routers is not included in this release.

This command was modified. The ds0 and dsp keywords were added.

Usage Guidelines

The show call resource voice stats command displays the H.323 resources that are monitored when the

resource threshold command is used to configure resource threshold reporting.

Examples

The following is sample output from the show call resource voice stats command, which shows the resource statistics for an H.323 gateway:

Router#

show call resource voice stats

Resource Monitor Dial-up Resource Statistics Information:

DSP Statistics:

Utilization: 0 percent

Total channels: 48

Inuse channels: 0

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Disabled channels 0:

Pending channels: 0

Free channels: 48

DS0 Statistics:

Total channels: 0

Addressable channels: 0

Inuse channels: 0

Disabled channels: 0

Free channels: 0

The table below describes significant fields shown in this output.

Table 42: show call resource voice stats Field Descriptions

Statistic

Total channels

Inuse channels

Disabled channels

Pending channels

Free channels

Addressable channels

Definition

Number of channels physically configured for the resource.

Number of addressable channels that are in use. This value includes all channels that either have active calls or have been reserved for testing.

Number of addressable channels that are physically down or that have been disabled administratively with the shutdown or busyout command.

Number of addressable channels that are pending in loadware download.

Number of addressable channels that are free.

Number of channels that can be used for a specific type of dialup service, such as H.323, which includes all the DS0 resources that have been associated with a voice plain old telephone service (POTS) dial plan profile.

Related Commands

Command resource threshold show call resource voice threshold

Description

Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.

Displays the threshold configuration settings and status for an H.323 gateway.

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show call history fax through show debug condition show call resource voice threshold

show call resource voice threshold

To display the threshold configuration settings and status for an H.323 gateway, use the show call resource

voice threshold command in privileged EXEC mode.

show call resource voice threshold [ds0| dsp]

Syntax Description ds0 dsp

(Optional) Specifies the voice digital signal level zero

(DS0) resource statistics information.

(Optional) Specifies the voice digital signal processor

(DSP) resource statistics information.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)T

12.1(5)XM2

12.2(2)XB1

12.4(22)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.1(5)XM2

This command was integrated into Cisco IOS Release into 12.2(2)XB1.

This command was modified. The ds0 and dsp keywords were added.

Usage Guidelines

The show call resource voice threshold command displays the H.323 resource thresholds that are configured with the resource threshold command.

Examples

The following is sample output from the show call resource voice threshold command, which shows the resource threshold settings and status for an H.323 gateway:

Router#

show call resource voice threshold

Resource Monitor Dial-up Resource Threshold Information:

DS0 Threshold:

Client Type: h323

High Water Mark: 70

Low Water Mark: 60

Threshold State: init

DSP Threshold:

Client Type: h323

High Water Mark: 70

Low Water Mark: 60

Threshold State: low_threshold_hit

The table below describes the significant fields shown in the display.

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Table 43: show call resource voice threshold Field Descriptions

Field

High Water Mark

Low Water Mark

Description

Resource-utilization level that triggers a message indicating that H.323 resource use is high. The range is 1 to 100. A value of 100 indicates that the resource is unavailable.The default is 90.

Resource-utilization level that triggers a message indicating that H.323 resource use has dropped below the high-usage level. The range is 1 to 100. The default is 90.

Related Commands

Command resource threshold show call resource voice stats

Description

Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.

Displays resource statistics for an H.323 gateway.

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show call history fax through show debug condition show call rsvp-sync conf

show call rsvp-sync conf

To display the configuration settings for Resource Reservation Protocol (RSVP) synchronization, use the

show call rsvp-sync confcommand in privileged EXEC mode.

show call rsvp-sync conf

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(3)XI1

12.1(5)T

12.2(2)XB1

12.2(8)T

12.2(11)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series,

Cisco 7200, Cisco MC3810, Cisco AS5300, and Cisco AS5800.

This command was integrated into Cisco IOS Release 12.1(5)T.

This command was implemented on the Cisco AS5850.

Support for the Cisco AS5300,Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 is not included in this release.

This command is supported on the Cisco AS5300, Cisco AS5350, Cisco

AS5400, Cisco AS5800, and Cisco AS5850 in this release.

Examples

The following example shows sample output from this command:

Router#

show call rsvp-sync conf

VoIP QoS: RSVP/Voice Signaling Synchronization config:

Overture Synchronization is ON

Reservation Timer is set to 10 seconds

The table below describes significant fields shown in this output.

Table 44: show call rsvp-sync conf Field Descriptions

Field

Overture Synchronization is ON

Reservation Timer is set to xx seconds

Description

Indicates whether RSVP synchronization is enabled.

Number of seconds for which the RSVP reservation timer is configured.

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Related Commands

Command

call rsvp -sync

call rsvp -sync resv-timer

debug call rsvp -sync events

show call rsvp -sync stats

show call rsvp-sync conf

Description

Enables synchronization between RSVP and the

H.323 voice signaling protocol.

Sets the timer for RSVP reservation setup.

Displays the events that occur during RSVP synchronization.

Displays statistics for calls that attempted RSVP reservation.

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show call history fax through show debug condition show call rsvp-sync stats

show call rsvp-sync stats

To display statistics for calls that attempted Resource Reservation Protocol (RSVP) reservation, use the show

call rsvp-sync stats command in privileged EXEC mode.

show call rsvp-sync stats

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(3)XI1

12.1(5)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.1(5)T.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Examples

The following example shows sample output from this command:

Router#

show call rsvp-sync stats

VoIP QoS:Statistics Information:

Number of calls for which QoS was initiated : 18478

Number of calls for which QoS was torn down : 18478

Number of calls for which Reservation Success was notified : 0

Total Number of PATH Errors encountered : 0

Total Number of RESV Errors encountered : 0

Total Number of Reservation Timeouts encountered : 0

The table below describes significant fields shown in this output.

Table 45: show call rsvp-sync stats Field Descriptions

Field

Number of calls for which QoS was initiated

Number of calls for which QoS was torn down

Description

Number of calls for which RSVP setup was attempted.

Number of calls for which an established RSVP reservation was released.

Number of calls for which Reservation Success was notified

Number of calls for which an RSVP reservation was successfully established.

Total Number of PATH Errors encountered Number of path errors that occurred.

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Field

Total Number of RESV Errors encountered

Description

Number of reservation errors that occurred.

Total Number of Reservation Timeouts encountered Number of calls in which the reservation setup was not complete before the reservation timer expired.

Related Commands

Command

call rsvp -sync

call rsvp -sync resv-timer

debug call rsvp -sync events

show call rsvp -sync conf

Description

Enables synchronization between RSVP and the

H.323 voice signaling protocol.

Sets the timer for RSVP reservation setup.

Displays the events that occur during RSVP synchronization.

Displays the RSVP synchronization configuration.

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show call history fax through show debug condition show call spike status

show call spike status

To display the configured call spike threshold and statistics for incoming calls, use the show call spike status command in privileged EXEC mode.

show call spike status [dial-peer tag]

Syntax Description dial-peer

tag

(Optional) Displays configuration information for a dial peer.

(Optional) Specifies the dial peer identifying number.

Range is from 1 to 2147483647.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(4)XM

12.2(8)T

12.2(11)T

15.1(3)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(4)T. This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.

This command was implemented on the Cisco AS5850.

This command was implemented on the Cisco 1750 and Cisco 1751. This command was not supported on any other platforms in this release.

This command was implemented on the Cisco 7200 series. Support for the Cisco

AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was not included in this release.

Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was added in this release.

This command was modified. The output fields of the command were modified to include the output at the dial peer level.

Examples

The following is sample output from this command:

Router#

show call spike status

Call Spiking:Configured

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show call history fax through show debug condition show call spike status

Call spiking :NOT TRIGGERED total call count in sliding window ::20

The table below describes the significant fields shown in the display.

Table 46: show call spike status Field Descriptions

Field

Call Spiking

Call Spiking total call count in sliding window

Description

Current enabled state of call spiking.

Details if the call spiking limit has been triggered.

Number of calls during the spiking interval.

Router#

show call spike status dial-peer 400

TAG CONFIG SPIKED TOTAL REJECTED CALLS

400 YES NO 4

REJECTED CALLS

0

The table below describes the significant fields shown in the display.

Table 47: show call spike status (dial peer) Field Descriptions

Field

TAG

CONFIG

SPIKED

TOTAL REJECTED CALLS

REJECTED CALLS

Description

Dial peer tag.

Displays if the call spike command has been configured.

Details if the call spiking limit has been triggered.

Displays the number of calls rejected due to a call spike in the dial peer.

Displays the number of calls rejected when the call spike was triggered until the call spike control was released.

Related Commands

Command call spike

Description

Configures the limit for the number of incoming calls in a short period of time.

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show call threshold

To display enabled triggers, current values for configured triggers, and the number of application programming interface (API) calls that were made to global and interface resources, use the show call threshold command in privileged EXEC mode.

show call threshold {config| status [unavailable]| stats}

Syntax Description config status unavailable stats

Displays the current threshold configuration.

Displays the status of all configured triggers and whether or not the CPU is available.

(Optional) Displays the status for all unavailable resources.

Displays statistics for API calls; that is, the resource-based measurement.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(4)XM

12.2(8)T

12.2(11)T

15.2(2)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(4)T. This command is not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms in this release.

This command was implemented on the Cisco AS5850.

This command was implemented on the Cisco 1750 and Cisco 1751. This command is not supported on any other platforms in this release.

This command was implemented on the Cisco 7200 series. Support for the Cisco

AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco

AS5400, and Cisco AS5850.

This command was modified. The output was modified to display the configured bandwidth threshold, bandwidth availability, and call admission control statistics.

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Examples

The following is sample output from the show call threshold config command:

Router#

show call threshold config

Some resource polling interval:

IF

CPU_AVG interval: 60

Memory interval: 5

Type

-----

Serial3/1:23

N/A

---int-calls cpu-avg

Value Low

--------

0 107

0 70

High

----

107

90

Enable

------

N/A busy&treat

The following is sample output from the show call threshold status command:

Router#

show call threshold status

Status IF

--------

Avail N/A

Avail N/A

Type

-----total-calls cpu-avg

Value Low

-------

0

0

5

5

High

----

5000

65

Enable

----busyout busyout

The following is sample output from the show call threshold status unavailable command:

Router#

show call threshold status unavailable

Unavailable configured resources at the current time:

IF Type Value Low High

------------------

Enable

-----

The following is sample output from the show call threshold stats command:

Router#

show call threshold stats

Total resource check: 0 successful: 0 failed: 0

The table below describes significant fields shown in this output.

Table 48: show call threshold Field Descriptions

Field

CPU_AVG interval

Memory interval

IF

Type

Value

Low

High

Enable

Description

Interval of configured trigger CPU_AVG.

Interval of configured trigger Memory.

Interface type and number.

Type of resource.

Value of a call that is to be matched against low and high thresholds.

Low threshold.

High threshold.

Shows if busyout and the call treatment command are enabled.

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show call threshold

Related Commands

Command call threshold call threshold poll-interval clear call threshold show call history fax through show debug condition

Description

Enables a resource and defines associated parameters.

Enables a polling interval threshold for CPU or memory.

Clears enabled triggers and their associated parameters.

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show call treatment

To display the call-treatment configuration and statistics for handling the calls on the basis of resource availability, use the show call treatment command in privileged EXEC mode.

show call treatment {config| stats}

Syntax Description config stats

Displays the call treatment configuration.

Displays statistics for handling the calls on the basis of resource availability.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(4)XM

12.2(8)T

12.2(11)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(4)T. This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.

This command was implemented on the Cisco AS5850.

This command was implemented on the Cisco 1750 and Cisco 1751. This command was not supported on any other platforms in this release.

This command was implemented on the Cisco 7200 series. Support for the Cisco

AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

Examples

The following is sample output from thiscommand:

Router#

show call treatment config

Call Treatment Config

---------------------

Call treatment is OFF.

Call treatment action is: Reject

Call treatment disconnect cause is: no-resource

Call treatment ISDN reject cause-code is: 41

The table below describes significant fields shown in this output.

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Table 49: show call treatment config Field Descriptions

Field

Call treatment is:

Call treatment action is:

Call treatment disconnect cause is:

Call treatment ISDN reject cause-code is:

Description

State of call treatment, either ON or OFF.

Action trigger assigned for call treatment.

Reason for disconnect.

Reject code number assigned.

The following is sample output from the show call treatmentcommand:

Router# show call treatment stats

Call Treatment Statistics

-------------------------

Total Calls by call treatment: 0

Calls accepted by call treatment: 0

Calls rejected by call treatment: 0

Reason

-----cpu-5sec: cpu-avg: total-mem: io-mem: proc-mem: total-calls:

Num. of calls rejected

----------------------

0

0

0

0

0

0

The table below describes significant fields shown in this output.

Table 50: show call treatment stats Field Descriptions

Field

Total Calls by call treatment:

Calls accepted by call treatment:

Calls rejected by call treatment: cpu-5sec cpu-avg total-mem io-mem proc-mem

Description

Number of calls received and treated.

Calls that passed treatment parameters.

Calls that failed treatment parameters.

Number of calls rejected for failing the cpu-5sec parameter.

Number of calls rejected for failing the cpu-avg parameter.

Number of calls rejected for failing the total-mem parameter.

Number of calls rejected for failing the io-mem parameter.

Number of calls rejected for failing the proc-mem parameter.

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Field

total-calls

Related Commands

Command call treatment on call treatment action call treatment cause-code call treatment isdn-reject clear call treatment stats show call treatment

Description

Number of calls rejected for failing the total-calls parameter.

Description

Enables call treatment to process calls when local resources are unavailable.

Configures the action that the router takes when local resources are unavailable.

Specifies the reason for the disconnection to the caller when local resources are unavailable.

Specifies the rejection cause-code for ISDN calls when local resources are unavailable.

Clears the call-treatment statistics.

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show call history fax through show debug condition show call-router routes

show call-router routes

To display the routes cached in the current border element (BE), use the show call-router routes in EXEC mode.

show call-router routes [static| dynamic| all]

Syntax Description static dynamic all

Descriptors provisioned on the border element.

Dynamically learned descriptors.

Both static and dynamic descriptors.

Command Default

All

Command Modes

EXEC (#)

Command History

Release

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(11)T

Examples

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(4)T.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

The following example is sample output from this command.

Router#

show call-router routes

Static Routes:

==============

DescriptorID= 6561676C65000000000000000000000A lastChanged = 19930301063311

IP addr :port

172.18.195.64

:2099

Dynamic Routes:

Prefix

5553122

===============

DescriptorID= 506174726F6E6F757300000000000002 lastChanged = 19930228190012

IP addr :port

172.18.195.65

:2099

Prefix

310

DescriptorID= 506174726F6E6F757300000000000003 lastChanged = 19930228190012

IP addr :port

172.18.195.65

:2099

Prefix

555301

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DescriptorID= 506174726F6E6F757300000000000004 lastChanged = 19930228190012

IP addr :port

172.18.195.65

:2099

Prefix

555302

DescriptorID= 506174726F6E6F757300000000000005 lastChanged = 19930228190012

IP addr :port

172.18.195.65

:2099

Prefix

818

DescriptorID= 506174726F6E6F757300000000000001 lastChanged = 19930228190012

IP addr :port

172.18.195.65

:2099

Prefix

1005

Field descriptions should be self-explanatory.

Related Commands

Command show call-router active show call-router history show call-router status show dial-peer voice show num-exp show voice port

Description

Displays active call information for a voice call in progress.

Displays the VoIP call-history table.

Displays the Annex G BE status.

Displays configuration information for dial peers.

Displays how the number expansions are configured in VoIP.

Displays configuration information about a specific voice port.

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show call-router status

To display the Annex G border element status, use the show call-router statuscommand in user EXEC mode.

show call-router status [neighbors]

Syntax Description neighbors

(Optional) Displays the neighbor border element status.

Command Modes

User EXEC (#)

Command History

Examples

Release

12.2(2)XA

12.2(4)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(4)T.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T and modified to add the neighborskeyword.

The following example displays the Annex G border element status. Note that the example shows the status for two neighbors.:

Router#

show call-router status neighbors

ANNEX-G CALL ROUTER STATUS:

===========================

Border Element ID Tag : Celine

Domain Name

Border Element State

: Celine-Domain

: UP

Border Element Local IP : 172.18.193.31:2099

Advertise Policy : STATIC descriptors

Hopcount Value

Descriptor TTL

: 7

: 3180

Access Policy

Current Active Calls

: Neighbors only

: 0

Current Calls in Cache : 0

Cumulative Active Calls : 0

Usage Ind Messages Sent : 0

Usage Ind Cfm Rcvd

IRRs Received

: 0

: 0

DRQs Received : 0

Usage Ind Send Retrys : 0

NEIGHBOR INFORMATION:

=====================

Local Neighbor ID : (none)

Remote Element ID : (unknown)

Remote Domain ID : (unknown)

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IP Addr

Status

: 1.2.3.4:2099

: DOWN

Caching

Query Interval

: OFF

: 30 MIN (querying disabled)

Usage Indications :

Current Active Calls : 0

Retry Period : 600 SEC

Retry Window : 3600 MIN

Service Relationship Status: ACTIVE

Inbound Service Relationship : DOWN

Service ID : (none)

TTL : 1200 SEC

Outbound Service Relationship : DOWN

Service ID : (none)

TTL : (none)

Retry interval : 120 SEC (0 until next attempt)

The table below describes significant fields shown in this output.

Table 51: show call-router status Field Descriptions

Field

Border Element ID Tag

Border Element State

Border Element Local IP

Advertise Policy

Hopcount Value

Descriptor TTL

Access Policy

Local Neighbor ID

Service Relationship Status

Inbound Service Relationship

Outbound Service Relationship

Description

Identifier for the border element.

Indicates if the border element is running.

Local IP address of the border element.

Type of descriptors that the border element advertises to its neighbors. Default is static. Other values are

dynamic and all.

Maximum number of border element hops through which an address resolution request can be forwarded.

Default is 7.

Time-to-live value, or the amount of time, in seconds, for which a route from a neighbor is considered valid.

Range is from 1 to 2147483647. Default is 1800 (30 minutes).

Requires that a neighbor be explicitly configured for requests to be accepted.

Domain name reported in service relationships.

Service relationship between two border elements is active.

Inbound time-to-Live (TTL) value in number of seconds. Range is from 1 to 4294967295.

Specifies the amount of time, in seconds, to establish the outbound relationship. Range is from 1 to 65535.

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Field

Retry interval

Related Commands

Command advertise

call -router

hopcount local shutdown ttl show call history fax through show debug condition

Description

Retry value between delivery attempts, in number of seconds. Range is from 1 to 3600.

Description

Controls the type of descriptors that the border element advertises to its neighbors.

Enables the Annex G border element configuration commands.

Specifies the maximum number of border element hops through which an address resolution request can be forwarded.

Defines the local domain, including the IP address and port border elements that the border element should use for interacting with remote border elements.

Shuts down the Annex G border element.

Sets the expiration timer for advertisements.

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show ccm-manager

To display a list of Cisco CallManager servers and their current status and availability, use the show

ccm-manager command in privileged EXEC mode.

show ccm-manager [backhaul| config-download| fallback-mgcp| hosts| music-on-hold| redundancy|

download-tones [c1| c2]]

Syntax Description backhaul config-download fallback-mgcp hosts music-on-hold redundancy

download-tones c1 | c2

(Optional) Displays information about the backhaul link.

(Optional) Displays information about the status of

Media Gateway Control Protocol (MGCP) and Skinny

Client Control Protocol (SCCP) configuration download.

(Optional) Displays the status of the MGCP gateway fallback feature.

(Optional) Displays a list of each configured Cisco

CallManager server in the network, together with its operational status and host IP address.

(Optional) Displays information about all the multicast music-on-hold (MOH) sessions in the gateway at any given point in time.

(Optional) Displays failover mode and status information for hosts, including the redundant link port, failover interval, keepalive interval, MGCP traffic time, switchover time, and switchback mode.

(Optional) Displays custom tones downloaded to the gateway. The custom tone value of c1 or c2 specifies which tone information to display.

Command Default

If none of the optional keywords is specified, information related to all keywords is displayed.

Command Modes

Privileged EXEC (#)

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Command History

Release

12.1(3)T

12.2(2)XA

12.2(2)XN

12.2(11)T

12.2(15)ZJ

12.3(4)T

12.3(14)T

12.4(15)XY

Modification

This command was introduced on the Cisco CallManager Version 3.0 and Cisco

VG200.

This command was implemented on the Cisco 2600 series and Cisco 3600 series.

This command was modified to provide enhanced MGCP voice gateway interoperability to Cisco CallManager Version 3.1 for the Cisco 2600 series,

Cisco 3600 series, and Cisco VG200.

This command was integrated into Cisco IOS Release 12.2(11) and the Cisco

CallManager Version 3.2. It was implemented on the Cisco IAD2420 series.

The download-tones [ c1 | c2 ] keywords were added for the following platforms:

Cisco 2610XM, Cisco 2611XM, Cisco 2620XM, Cisco 2621XM, Cisco 2650XM,

Cisco 2651XM, Cisco 2691, Cisco 3640A, Cisco 3660, Cisco 3725, and Cisco

3745.

The keywords were integrated into Cisco IOS Release 12.3(4)T.

New output was added relating to SCCP autoconfiguration.

The display output was modified to include the number of TFTP download failures allowed.

Usage Guidelines

Use the show ccm-manager config-download command to determine the status of Cisco Unified

Communications Manager servers and the automatic download information and statistics.

Examples

The following sample output shows the configured amplitudes, frequencies, and cadences of custom tone 1,

Hong Kong:

Router#

show ccm-manager download-tones c1

!

Custom Tone 1 : Hong Kong

Pulse dial:normal, Percent make:35%, DTMF low Amp.= 65424, high Amp.= 65446, Pcm:u-Law

FXS FXO E&M FXS FXO E&M

Dual Tone DR NF FOF FOS AOF AOF AOF AOS AOS AOS ONTF OFTF ONTS OFTS ONTT OFTT ONT4 OFT4

(optional) FOF2 FOS2 FOF3 FOS3 FOF4 FOS4 FOT FO4 AOT AO4 RCT1 RCT2 RCT3 RCT4

BUSY 0 2 480 620 -120 -120 -120 -120 -120 -120 500 500 0 0 0 0 0 0

RING_BACK 0 2 440 520 -120 -120 -120 -120 -120 -120 400 200 400 3000

CONGESTION 0 2 480 620 -200 -200 -200 -240 -240 -240 250 250 0 0

NUMBER_UNOBTAINABLE 0 2 480 620 -120 -120 -120 -120 -120 -120 65535

DIAL_TONE 0 2 350 440 -150 -150 -150 -150 -150 -150 65535

DIAL_TONE2

OUT_OF_SERVICE

0

0

2

1

350

950

440 -150 -150 -150 -150 -150 -150 65535

0 -150 -150 -150 0 0 0 330

0

330

0

0

0

0

0

0

0

0

0

0

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ADDR_ACK

DISCONNECT

OFF_HOOK_NOTICE

OFF_HOOK_ALERT

0 1 600 0 -240 -240 -240 0 0 0 125 125 125 65535

0 1 600 0 -150 -150 -150 0 0 0 330 330 330 65535

0 2 1400 2040 -240 -240 -240 -240 -240 -240 100 100 0 0

0 2 1400 2040 -240 -240 -240 -240 -240 -240 100 100 0 0

WAITING

CONFIRM

CNFWRN_J

CNFWRN_D

0 0

0 0

0 1 950

0 1 600

0

0

0

0

0

0

0

0

0 -170 -170 -190

0 -170 -170 -190

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0 100 100 100 65535

0 100 100 100 65535

0

0

STUTT_DIALTONE

100 100 65535

PERM_SIG_TONE

WAITING1

0 2 350 440 -150 -150 -150 -150 -150 -150 100 100 100 100

0

0 1 480

0 0 0

0 -170 -170 -170

0 0 0 0

0

0

0

0

0 65535

0 0

0

0

0

0

0

0

WAITING2 0 0 0 0 0 0 0 0 0 0 0 0 0 0

WAITING3

WAITING4

0 0

0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

MSGWAIT_IND

OFF_HOOK_WARN

0 0

0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

Sequence Tone DR NF F1C1 F2C1 AOF AOS C1ONT C1OFT C2ONT C2OFT C3ONT C3OFT C4ONT

C4OFT F1C2 F2C2 F1C3 F2C3 F1C4 F2C4

INTERCEPT

TONE_ON_HOLD

NO_CIRCUIT

Legend:

0 0

0 0

0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

DR: direction NF: number of frequency FO<F,S,T,4>: frequency of<1st,2nd,3rd,4th> AO<F,S,T,4>: amplitude of<1st,2nd,3rd,4th>

FOF<1-4>: frequency of 1st, cadence<1-4> FOS<1-4>: frequency of 2nd, cadence<1-4>

RCT<1-4>: repeat count for cadence<l-4> F(1-4>C<1-4> : frequency<1-4> of cadence<1-4>

C<1-4>ONT: cadence<1-4> on time C<1-4>OFT: cadence<1-4> off time

The three tables below and give descriptions of significant fields once the tones are automatically downloaded to the gateway.

0

0

Table 52: show ccm-manager download-tones Significant Output Fields

Field

Percent make

DTMF low Amp.

high Amp.

Pcm

Description

Pulse ratio in percentage of make.

Low frequency level.

High frequency level.

Pulse Code Modulation (mu-law or a-law).

Table 53: show ccm-manager download-tones Output Fields for Dual Tones

Field of Dual Tone

DR

Description

Direction to PSTN (0) or Packet Network (1).

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ONTT

OFTT

ONT4

OFT4

FOF2

Field of Dual Tone

NF

FOF

FXS AOF

FXO AOF

E&M AOF

FXS AOS

FXO AOS

E&M AOS

ONTF

OFTF

ONTS

OFTS

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Description

Number of Frequency (from 1 to 4).

Frequency of First component (in Hz).

Amplitude of First component (from 1 to 65535 = +3 dBm0) for the foreign exchange station (FXS).

Amplitude of First component (from 1 to 65535 = +3 dBm0) for the foreign exchange office (FXO).

Amplitude of First component (from 1 to 65535 = +3 dBm0) for the recEive and transMit (E&M).

Amplitude of Second component (from 1 to 65535 =

+3 dBm0) for the FXS.

Amplitude of Second component (from 1 to 65535 =

+3 dBm0) for the FXO.

Amplitude of Second component (from 1 to 65535 =

+3 dBm0) for the E&M.

On time; time the tone is generated (milliseconds) for the first frequency.

Off time; silence time (milliseconds) for the first frequency.

On time; time the tone is generated (milliseconds) for the second frequency.

Off time; silence time (milliseconds) for the second frequency.

On time; time the tone is generated (milliseconds) for the third frequency.

Off time; silence time (milliseconds) for the third frequency.

On time; time the tone is generated (milliseconds) for the fourth frequency.

Off time; silence time (milliseconds) for the fourth frequency.

Frequency of First component for the second cadence.

show call history fax through show debug condition show ccm-manager

FOT

FO4

AOT

AO4

Field of Dual Tone

FOS2

FOF3

FOS3

FOF4

FOS4

RCT1

RCT2

RCT3

RCT4

Description

Frequency of Second component for the second cadence.

Frequency of First component for the third cadence.

Frequency of Second component for the third cadence.

Frequency of First component for the fourth cadence.

Frequency of Second component for the fourth cadence.

Frequency of Third component (in Hertz).

Frequency of Fourth component (in Hertz).

Amplitude of Third component (from 1 to 65535 =

+3 dBm0).

Amplitude of Fourth component (from 1 to 65535 =

+3 dBm0).

Number of repeat for the first cadence.

Number of repeat for the second cadence.

Number of repeat for the third cadence.

Number of repeat for the fourth cadence.

Table 54: show ccm-manager download-tones Output Fields for Sequence Tones

Field of Sequence Tone

DR

NF

F1C1

F2C1

AOF

AOS

Description

Direction to PSTN (0) or Packet Network (1).

Number of Frequency (from 1 to 4).

Frequency 1 of Cadence 1.

Frequency 2 of Cadence 1.

Amplitude of First component (from 1 to 65535).

Amplitude of Second component (from 1 to 65535).

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F2C2

F1C3

F2C3

F1C4

F2C4

Field of Sequence Tone

C1ONT

C1OFT

C2ONT

C2OFT

C3ONT

C3OFT

C4ONT

C4OFT

F1C2

Description

Cadence 1 On Time.

Cadence 1 Off Time.

Cadence 2 On Time.

Cadence 2 Off Time.

Cadence 3 On Time.

Cadence 3 Off Time.

Cadence 4 On Time.

Cadence 4 Off Time.

Frequency 1 of Cadence 2.

Frequency 2 of Cadence 2.

Frequency 1 of Cadence 3.

Frequency 2 of Cadence 3.

Frequency 1 of Cadence 4.

Frequency 2 of Cadence 4.

The following is sample output from the show ccm-manager command for displaying the status and availability of both the primary and the backup Cisco Unified Communications Manager server:

Router# show ccm-manager

MGCP Domain Name: Router2821.cisco.com

Priority Status Host

============================================================

Primary

First Backup

Registered

None

10.78.236.222

Second Backup None

Current active Call Manager: 10.78.236.222

Backhaul/Redundant link port: 2428

Failover Interval: 30 seconds

Keepalive Interval: 15 seconds

Last keepalive sent:

Last MGCP traffic time:

Last failover time:

Last switchback time:

Switchback mode:

MGCP Fallback mode:

Last MGCP Fallback start time: None

Last MGCP Fallback end time: None

MGCP Download Tones: Disabled

TFTP retry count to shut Ports: 3

PRI Backhaul Link info:

Link Protocol: TCP

21:48:37 UTC Nov 4 2007 (elapsed time: 00:00:15)

21:48:51 UTC Nov 4 2007 (elapsed time: 00:00:02)

None

None

Graceful

Not Selected

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Remote Port Number: 2428

Remote IP Address: 172.20.71.38

Current Link State: OPEN

Statistics:

Packets recvd: 1

Recv failures: 0

Packets xmitted: 3

Xmit failures: 0

PRI Ports being backhauled:

Slot 1, port 1

MGCP Download Tones: Enabled

Configuration Auto-Download Information

=======================================

Current version-id: {1645327B-F59A-4417-8E01-7312C61216AE}

Last config-downloaded:00:00:49

Current state: Waiting for commands

Configuration Download statistics:

Download Attempted

Download Successful

Download Failed

: 6

: 6

: 0

Configuration Attempted

Configuration Successful

: 1

: 1

Configuration Failed(Parsing): 0

Configuration Failed(config) : 0

Last config download command: New Registration

Configuration Error History:

FAX mode: cisco

The table below describes the significant fields shown in the display.

Table 55: show ccm-manager Field Descriptions

Field

MGCP Domain Name (system)

Priority

Status

Host

Current active Call Manager

Backhaul/Redundant link port

Failover Interval

Description

System used in the Internet for translating domain names of network nodes into IP addresses.

Priority of the Cisco CallManager servers present in the network. Possible priorities are primary, first backup, and second backup.

Current usage of the Cisco Unified Communications

Manager server. Values are Registered, Idle, Backup

Polling, and Undefined.

Host IP address of the Cisco CallManager server.

IP address of the active Cisco Communications

Manager server. This field can be the IP address of any one of the following Cisco Communications

Manager servers: Primary, First Backup, and Second

Backup.

Port that the Cisco CallManager server is to use.

Maximum amount of time that can elapse without the gateway receiving messages from the currently active

Cisco Call Manager before the gateway switches to the backup Cisco Call Manager.

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Field

Keepalive Interval

Last keepalive sent

Last MGCP traffic time

Switchback mode

MGCP Fallback mode

Last MGCP Fallback start time

Lasts MGCP Fallback end time

MGCP Download Tones

TFTP retry count to shut Ports

Description

Interval following which, if the gateway has not received any messages from the currently active Cisco

Communications Manager server within the specified amount of time, the gateway sends a keepalive message to the Cisco Communications Manager server to determine if it is operational.

Time in hours (military format), minutes and seconds at which the last keepalive message was sent.

Time in hours (military format), minutes and seconds at which the last MGCP traffic message was sent.

Displays the switchback mode configuration that determines when the primary Cisco CallManager server is used if it becomes available again while a backup Cisco CallManager server is being used.

Values that can appear in this field are Graceful,

Immediate, Schedule -time, and Uptime-delay.

Displays the MGCP fallback mode configuration. If

"Not Selected" displays, then fallback is not configured. If "Enabled/OFF" displays, then fallback is configured but not in effect. If "Enabled/ON" displays, then fallback is configured and in effect.

Start time stamp in hours (military format), minutes and seconds of the last fallback.

End time stamp in hours (military format), minutes and seconds of the last fallback.

Displays if the customized tone download is enabled.

Number of TFTP download failures allowed before endpoints are shutdown.

The following is sample output from the show ccm-manager config-downloadcommand showing the status of the SCCP download:

Router#

show ccm-manager config-download

Configuration Auto-Download Information

=======================================

Current version-id:{4171F93A-D8FC-49D8-B1C4-CE33FA8095BF}

Last config-downloaded:00:00:47

Current state:Waiting for commands

Configuration Download statistics:

Download Attempted

Download Successful

Download Failed

:6

:6

:0

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Configuration Attempted

Configuration Successful

:1

:1

Configuration Failed(Parsing):0

Configuration Failed(config) :0

Last config download command:New Registration

SCCP auto-configuration status

===============================================================

Registered with Call Manager: No

Local interface: FastEthernet0/0 (000c.8522.6910)

Current version-id: {D3A886A2-9BC9-41F8-9DB2-0E565CF51E5A}

Current config applied at: 04:44:45 EST Jan 9 2003

Gateway downloads succeeded: 1

Gateway download attempts: 1

Last gateway download attempt: 04:44:45 EST Jan 9 2003

Last successful gateway download: 04:44:45 EST Jan 9 2003

Current TFTP server: 10.2.6.101

Gateway resets: 0

Gateway restarts: 0

Managed endpoints: 6

Endpoint downloads succeeded: 6

Endpoint download attempts: 6

Last endpoint download attempt: 04:44:45 EST Jan 9 2003

Last successful endpoint download: 04:44:45 EST Jan 9 2003

Endpoint resets: 0

Endpoint restarts: 0

Configuration Error History: sccp ccm CCM-PUB7 identifier 1 end controller T1 2/0no shut controller T1 2/0no shut controller T1 2/0no shut isdn switch-type primary-ni end

The table below describes the significant fields shown in the display.

Table 56: show ccm-manager config-download Field Descriptions

Field

Current state

Download Attempted

Configuration Attempted

Managed endpoints

Endpoint downloads succeeded

Endpoint download attempts

Endpoint resets

Description

Current configuration state.

Number of times the gateway has tried to download the configuration file. The number of successes and failures is displayed.

Number of times the gateway has tried to configure the gateway based on the configuration file. The number of successes and failures is displayed.

Number of SSCP-controlled endpoints (analog and

BRI phones).

Number of times the gateway has successfully downloaded the configuration files for

SCCP-controlled endpoints.

Number of times the gateway has tried to download the configuration files for SCCP-controlled endpoints.

Number of SCCP gateway resets.

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Field

Endpoint restarts

Configuration Error History

Description

Number of SCCP gateway restarts.

Displays SCCP autoconfiguration errors.

The following is sample output from the show ccm-manager fallback-mgcp command:

Router#

show ccm-manager fallback-mgcp

Current active Call Manager: 172.20.71.38

MGCP Fallback mode: Enabled/OFF

Last MGCP Fallback start time: 00:14:35

Last MGCP Fallback end time: 00:17:25

The table below displays te mode. Modes are as follows:

Table 57: show ccm-manager fallback-mgcp modes

Field

MGCP Fallback mode

Description

The following are displayed:

• Not Selected--Fallback is not configured.

• Enabled/OFF--Fallback is configured but not in effect.

• Enabled/ON--Fallback is configured and in effect.

Last MGCP Fallback start time

Last MGCP Fallback end time

Start time stamp in hh:mm:ss of the last fallback.

End time stamp in hh:mm:ss of the last fallback.

The following is sample output from the show ccm-manager music-on-hold command:

Router#

show ccm-manager music-on-hold

Current active multicast sessions :1

Multicast

Address

RTP port number

Packets in/out

Call Codec id

Incoming

Interface

===================================================================

172.20.71.38

2428 5/5 99 g711

The table below describes the significant fields shown in the display.

Table 58: show ccm-manager music-on-hold Field Descriptions

Field

Current active multicast sessions

Multicast Address

Description

Number of active calls on hold.

Valid class D address from which the gateway is getting the RTP streams.

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Field

RTP port number

Packets in/out

Call id

Codec

Incoming Interface

Related Commands

Command ccm-manager config debug ccm-manager show ccm-manager show ccm-manager fallback-mgcp show isdn status show mgcp show ccm-manager

Description

Valid RTP port number on which the gateway receives the RTP packets.

Number of RTP packets received and sent to the digital signal processor (DSP).

Call ID of the call that is on hold.

Codec number.

Interface through which the gateway is receiving the

RTP stream.

Description

Supplies the local MGCP voice gateway with the IP address or logical name of the TFTP server from which to download XML configuration files and enable the download of the configuration.

Displays debugging information about the Cisco

CallManager.

Displays a list of Cisco CallManager servers, their current status, and their availability.

Displays the status of the MGCP gateway fallback feature.

Displays the Cisco IOS gateway ISDN interface status.

Displays the MGCP configuration information.

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show call history fax through show debug condition show cdapi

show cdapi

To display the Call Distributor Application Programming Interface (CDAPI), use the show cdapicommand in privileged EXEC mode.

show cdapi

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(7)T

12.3(4)T

Modification

This command was introduced on the Cisco AS5300.

This command was enhanced to display V.120 call types registering with the modem.

Usage Guidelines

CDAPI is the internal application programming interface (API) that provides an interface between signaling stacks and applications.

Examples

The following is sample output from the show cdapi command. The output displays the following information:

• Signaling stacks that register with CDAPI

• Applications that register with CDAPI

• Active calls

• Call type of each active call

• Message buffers in use

Router#

show cdapi

Registered CDAPI Applications/Stacks

====================================

Signaling Stack: ISDN

Interface: Se6/0:23

Application: TSP CDAPI Application Voice

Application Type(s) : Voice Data Facility Signaling V110 V120

Application Level : Tunnel

Application Mode : Enbloc

Application: TSP CDAPI Application COT

Application Type(s) : Cot

Application Level : Tunnel

Application Mode

Application: CSM

: Enbloc

Application Type(s) : Modem V110 V120

Application Level : Basic

Application Mode : Enbloc

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Signaling Stack: XCSP

Application: dialer

Application Type(s) : Data

Application Level : Basic

Application Mode : Enbloc

Active CDAPI Calls

==================

Se7/7:23 Call ID = 0x7717, Call Type = V.120, Application = CSM

CDAPI Message Buffers

=====================

Free Msg Buffers: 320

Free Raw Buffers: 320

Free Large-Raw Buffers: 120

Field descriptions should be self-explanatory. However, the following information may be of help:

• Enbloc is the mode where all call-establishment information is sent in the setup message (opposite of overlap mode, where additional messages are needed to establish the call).

• Cot is the Continuity Test (COT) subsystem that supports the continuity test required by the Signaling

System 7 (SS7) network to conduct loopback and tone check testing on the path before a circuit is established.

Related Commands

Command debug cdapi

Description

Displays information about the CDAPI.

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show call history fax through show debug condition show ces clock-select

show ces clock-select

To display the setting of the network clock for the specified port, use the show ces clock-selectcommand in privileged EXEC mode.

show ces slot/port clock-select

Syntax Description

slot

/port

Backplane slot number.

Interface port number. The slash must be entered.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(2)T

Modification

This command was introduced on the Cisco 3600 series.

Examples

The following is sample output from this command for slot 1, port 0:

Router#

show ces 1/0 clock-select

Priority 1 clock source:not configured

Priority 2 clock source:not configured

Priority 3 clock source:ATM1/0 UP

Priority 4 clock source:Local oscillator

Current clock source:ATM1/0, priority:3

Field descriptions should be self-explanatory.

Related Commands

Command clock-select

Description

Establishes the sources and priorities of the requisite clocking signals for the OC-3/STM-1 ATM Circuit

Emulation Service network module.

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show connect

To display configuration information about drop-and-insert connections that have been configured on a router, use the show connectcommand in privileged EXEC mode.

show connect {all| elements| name| id| port {T1| E1}slot/port}

Syntax Description all elements name id port

T1

E1

slot/port

Information for all configured connections.

Information for registered hardware or software interworking elements.

Information for a connection that has been named by using the connect global configuration command.

The name you enter is case sensitive and must match the configured name exactly.

Information for a connection that you specify by an identification number or range of identification numbers. The router assigns these IDs automatically in the order in which they were created, beginning with 1. The show connect allcommand displays these

IDs.

Information for a connection that you specify by indicating the type of controller (T1 or E1) and location of the interface.

T1 controller.

E1 controller.

Location of the T1 or E1 controller port whose connection status you want to see. Valid values for

slot and port are 0and 1. The slash must be entered.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

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Command History

Release

12.0(5)XK

12.0(7)T

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series.

This command was integrated into Cisco IOS Release 12.0(7)T.

Usage Guidelines

This command shows drop-and-insert connections on modular access routers that support drop-and-insert. It displays different information in different formats, depending on the keyword that you use.

Examples

The following examples show how the same tabular information appears when you enter different keywords:

Router#

show connect all

ID Name Segment 1 Segment 2 State

========================================================================

1

2

Test

Test2

-T1 1/0 01

-T1 1/0 03

-T1 1/1 02

-T1 1/1 04

ADMIN UP

ADMIN UP

Router#

show connect id 1-2

ID Name Segment 1 Segment 2 State

========================================================================

1

2

Test

Test2

-T1 1/0 01

-T1 1/0 03

-T1 1/1 02

-T1 1/1 04

ADMIN UP

ADMIN UP

Router#

show connect port t1 1/1

ID Name Segment 1 Segment 2 State

========================================================================

1

2

Test

Test2

-T1 1/0 01

-T1 1/0 03

-T1 1/1 02

-T1 1/1 04

ADMIN UP

ADMIN UP

The following examples show details about specific connections, including the number of time slots in use and the switching elements:

Router#

show connect id 2

Connection: 2 - Test2

Current State: ADMIN UP

Segment 1: -T1 1/0 03

TDM timeslots in use: 14-18 (5 total)

Segment 2: -T1 1/1 04

TDM timeslots in use: 14-18

Internal Switching Elements: VIC TDM Switch

Router#

show connect name Test

Connection: 1 - Test

Current State: ADMIN UP

Segment 1: -T1 1/0 01

TDM timeslots in use: 1-13 (13 total)

Segment 2: -T1 1/1 02

TDM timeslots in use: 1-13

Internal Switching Elements: VIC TDM Switch

Field descriptions should be self-explanatory.

Related Commands

Command connect

Description

Defines connections between T1 or E1 controller ports for Drop and Insert.

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Command tdm-group show connect

Description

Configures a list of time slots for creating clear channel groups (pass-through) for TDM cross-connect.

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show call history fax through show debug condition show controllers rs366

show controllers rs366

To display information about the RS-366 video interface on the video dialing module (VDM), use the show

controllers rs366command in privileged EXEC mode.

show controllers rs366 slot port

Syntax Description

slot port

Slot location of the VDM module. Valid entries are

1 or 2.

Port location of the EIA/TIA-366 interface in the

VDM module.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)XK

12.0(7)T

Modification

This command was introduced on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.0(7)T.

Examples

The following example displays information about the RS-366 controller:

Router#

show controllers rs366 0 1

RS366:driver is initialized in slot 1, port 0:

STATUS STATE LSR LCR ICSR EXT T1 T2 T3 T4 T5

0x02 0x01 0x00 0x50 0xE0 0x00 5000 5000 5000 20000 10000

Dial string:

121C

The table below describes significant fields shown in this output.

Table 59: show controllers rs366 Field Descriptions

Field

STATUS

STATE

LSR

Description

Last interrupt status.

Current state of the state machine.

Line status register of the VDM.

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Field

LCR

ICSR

EXT

T1 through T5

Dial string

show controllers rs366

Description

Line control register of the VDM.

Interrupt control and status register of the VDM.

Extended register of the VDM.

Timeouts 1 through 5 of the watchdog timer, in milliseconds.

Most recently dialed number collected by the driver.

0xC at the end of the string indicates the EON (end of number) character.

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show controllers timeslots

To display the channel-associated signaling (CAS) and ISDN PRI state in detail, use the show controllers timeslots command in privileged EXEC mode.

show controllers t1/e1 controller-number timeslotstimeslot-range

Syntax Description tl/e1

controller -number

timeslots timeslot -range

Controller number of CAS or ISDN PRI time slot.

Range is from 0 to 7.

Timeslot. E1 range is from 1 to 31. T1 range is from

1 to 24.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

10.0

12.1(3)T

12.1(5)T

Modification

This command was introduced.

The timeslots keyword was added.

This command was implemented on the Cisco AS5400.

Usage Guidelines

Use this command to display the CAS and ISDN PRI channel state in detail. The command shows whether the DS0 channels of a controller are in idle, in-service, maintenance, or busyout states. Use the show controllers

e1 or show controllers t1 command to display statistics about the E1 or T1 links.

Examples

The following example shows that the CAS state is enabled on the Cisco AS5300 with a T1 PRI card:

Router#

show controllers timeslots

T1 1 is up:

Loopback: NONE

DS0 Type Modem <-> Service

State

Channel

State

Rx

A B C D

Tx

A B C D

-----------------------------------------------------------------------------------------

1 cas-modem 1 in insvc connected 1 1 1 1 1 1 1 1

5

6

7

2

3

4 cas cas cas cas cas cas

-

-

-

-

-

-

-

-

-

-

-

insvc insvc insvc insvc insvc insvc idle idle idle idle idle idle

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0

0 0 0 0

0 0 0 0

0

0

0

0

0

0

0

0

0

0

0

0

0

0 0 0 0

0 0 0 0

0 0 0 0

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8

9 cas cas

10 cas

11 cas

12 cas

13 cas

14 cas

15 cas

16 cas

17 cas

18 cas

19 cas

20 cas

21 cas

22 unused

23 unused

24 unused

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

insvc insvc maint maint maint maint maint maint maint maint maint maint maint maint idle idle static-bo static-bo static-bo static-bo static-bo static-bo static-bo static-bo static-bo dynamic-bo dynamic-bo dynamic-bo

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

0 0 0 0

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

1 1 1 1

The following example shows that the ISDN PRI state is enabled on the Cisco AS5300 with a T1 PRI card:

T1 2 is up:

Loopback: NONE

DS0 Type Modem <-> Service Channel Rx Tx

State State A B C D A B C D

---------------------------------------------------------------------------

1 pri

2 pri

-

-

-

insvc insvc idle idle

3 pri

4 pri

5 pri

6 pri

7 pri

8 pri

9 pri

10 pri

11 pri

12 pri

13 pri

14 pri

15 pri

16 pri

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

insvc insvc insvc insvc insvc insvc insvc insvc insvc insvc insvc insvc insvc insvc idle idle idle idle idle idle idle idle idle idle idle idle idle idle

17 pri

18 pri

19 pri

20 pri

21 pri-modem

22 pri-modem

23 pri-digi

24 pri-sig

-

-

-

-

2

1

-

-

-

-

-

insvc insvc insvc insvc in insvc out insvc idle idle idle idle busy busy in insvc

outofsvc busy reserved

Field descriptions should be self-explanatory.

Related Commands

Command show controllers e1 show controllers t1

Description

Displays information about E1 links.

Displays information about T1 links.

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show controllers voice

To display information about voice-related hardware, use the show controllers voice command inprivileged

EXEC mode.

show controllers voice

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)XQ

Modification

This command was introduced on the Cisco 1750.

Usage Guidelines

This command displays interface status information that is specific to voice-related hardware, such as the registers of the TDM switch, the host port interface of the digital signal processor (DSP), and the DSP firmware versions. The information displayed is generally useful only for diagnostic tasks performed by technical support.

Examples

The following is sample output from this command:

Router#

show controllers voice

EPIC Switch registers:

STDA 0xFF STDB 0xFF SARA 0xAD SARB 0xFF SAXA 0xFF SAXB 0x0 STCR 0x3F

MFAIR 0x3F

STAR 0x65 OMDR 0xE2 VNSR 0x0 PMOD 0x4C PBNR 0xFF POFD 0xF0 POFU 0x18

PCSR 0x1 PICM 0x0 CMD1 0xA0 CMD2 0x70 CBNR 0xFF CTAR 0x2 CBSR 0x20 CSCR

0x0

DSP 0 Host Port Interface:

HPI Control Register 0x202

InterfaceStatus 0x2A MaxMessageSize 0x80

RxRingBufferSize 0x6 TxRingBufferSize 0x9 pInsertRx 0x4 pRemoveRx 0x4 pInsertTx 0x6 pRemoveTx 0x6

Rx Message 0: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 4AC7 5F08 91D1 0000 0000 7DF1 69E5 63E1 63E2

0020: 6E7C ED67 DE5D DB5C DC60 EC7E 6BE1 58D3 50CD 4DCE

0040: 50D2 5AE5 7868 DA52 CE4A C746 C647 C94B D25A EAF4

0060: 5DD7 4FCD 4ACA 4ACC 4FD3 5DE8 F769 DC58 D352 D253

0080: D65B E573 6CDF 59D3 4ECF 4FD0

Rx Message 1: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 1CDD 3E48 3B74 0000 0000 3437 3D4C F0C8 BBB5

0020: B2B3 B7BF D25B 4138 3331 3339 435F CFBD B6B2 B1B4

0040: BBC8 7E48 3B34 3131 363D 4FDE C3B9 B3B1 B3B8 C2DB

0060: 533F 3833 3235 3B48 71CC BDB7 B4B5 B8BF CF67 483D

0080: 3836 383C 455B DAC6 BDB9 B9BB

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Rx Message 2: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 4AC8 5F08 9221 0000 0000 54DA 61F5 EF60 DA53

0020: CF4F CD4E D256 DB63 FCEE 5FDA 55D1 50CF 4FD3 56D8

0040: 5DE1 6E7C EC60 DC59 D655 D456 D85D DF6A F4F4 69E2

0060: 5CDD 5BDC 5BDE 61E9 6DF1 FF76 F16D E96A E566 EA6A

0080: EB6F F16D EF79 F776 F5F5 73F0

Rx Message 3: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 1CDE 3E48 3BC4 0000 0000 C0CC EC54 453E 3C3C

0020: 3F47 56F3 D1C7 C1BF C0C6 CEE1 6752 4A46 4648 4E59

0040: 6FE4 D6CF CDCE D2DA E57E 675E 5B5B 5E62 6B76 FCF6

0060: F6FA 7D75 7373 7BF5 EAE1 DCDA DADD E6FE 6559 514D

0080: 4D4E 5563 EFD9 CDC8 C5C6 CAD1

Rx Message 4: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 4AC6 5F08 9181 0000 0000 DD5B DC5E E161 E468

0020: FAFD 6CE1 5AD3 53D1 53D7 61EC EA59 CF4A C644 C344

0040: CA4E D86C 60D0 48C2 3EBD 3CBD 3EC0 47CF 5976 DF4F

0060: C945 C242 C146 C94E D668 73DB 54CE 4DCC 4DCE 53DB

0080: 64F9 ED63 DC59 DA58 DC5D E46C

Rx Message 5: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 1CDC 3E48 3B24 0000 0000 5B5B 5D62 6A76 FCF5

0020: F5F9 7D78 7374 7CF5 EAE1 DDDA DBDD E7FE 6559 514E

0040: 4D4F 5663 EFD8 CDC8 C6C6 CAD1 E760 4E46 403F 4047

0060: 5173 D5C7 BFBC BCBE C5D4 6D4C 3F3B 3939 3D46 5ADB

0080: C5BC B7B6 B8BD C8E8 4F3F 3835

Tx Message 0: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 4AC6 5F08 9181 0000 003C DD5B DC5E E161 E468

0020: FAFD 6CE1 5AD3 53D1 53D7 61EC EA59 CF4A C644 C344

0040: CA4E D86C 60D0 48C2 3EBD 3CBD 3EC0 47CF 5976 DF4F

0060: C945 C242 C146 C94E D668 73DB 54CE 4DCC 4DCE 53DB

0080: 64F9 ED63 DC59 DA58 DC5D E46C

Tx Message 1: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 1CDC 3E48 3B24 0000 003C 5B5B 5D62 6A76 FCF5

0020: F5F9 7D78 7374 7CF5 EAE1 DDDA DBDD E7FE 6559 514E

0040: 4D4F 5663 EFD8 CDC8 C6C6 CAD1 E760 4E46 403F 4047

0060: 5173 D5C7 BFBC BCBE C5D4 6D4C 3F3B 3939 3D46 5ADB

0080: C5BC B7B6 B8BD C8E8 4F3F 3835

Tx Message 2: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 4AC7 5F08 91D1 0000 003C 7DF1 69E5 63E1 63E2

0020: 6E7C ED67 DE5D DB5C DC60 EC7E 6BE1 58D3 50CD 4DCE

0040: 50D2 5AE5 7868 DA52 CE4A C746 C647 C94B D25A EAF4

0060: 5DD7 4FCD 4ACA 4ACC 4FD3 5DE8 F769 DC58 D352 D253

0080: D65B E573 6CDF 59D3 4ECF 4FD0

Tx Message 3: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 1CDD 3E48 3B74 0000 003C 3437 3D4C F0C8 BBB5

0020: B2B3 B7BF D25B 4138 3331 3339 435F CFBD B6B2 B1B4

0040: BBC8 7E48 3B34 3131 363D 4FDE C3B9 B3B1 B3B8 C2DB

0060: 533F 3833 3235 3B48 71CC BDB7 B4B5 B8BF CF67 483D

0080: 3836 383C 455B DAC6 BDB9 B9BB

Tx Message 4: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 4AC8 5F08 9221 0000 003C 54DA 61F5 EF60 DA53

0020: CF4F CD4E D256 DB63 FCEE 5FDA 55D1 50CF 4FD3 56D8

0040: 5DE1 6E7C EC60 DC59 D655 D456 D85D DF6A F4F4 69E2

0060: 5CDD 5BDC 5BDE 61E9 6DF1 FF76 F16D E96A E566 EA6A

0080: EB6F F16D EF79 F776 F5F5 73F0

Tx Message 5: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 1CDE 3E48 3BC4 0000 003C C0CC EC54 453E 3C3C

0020: 3F47 56F3 D1C7 C1BF C0C6 CEE1 6752 4A46 4648 4E59

0040: 6FE4 D6CF CDCE D2DA E57E 675E 5B5B 5E62 6B76 FCF6

0060: F6FA 7D75 7373 7BF5 EAE1 DCDA DADD E6FE 6559 514D

0080: 4D4E 5563 EFD9 CDC8 C5C6 CAD1

Tx Message 6: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 1CDA 3E48 3A84 0000 003C E75F 4E46 403F 4147

Cisco IOS Voice Command Reference - S commands

401

show call history fax through show debug condition show controllers voice

0020: 5174 D5C7 BFBC BCBE C5D4 6C4C 3F3B 3939 3D46 5BDA

0040: C5BC B7B6 B8BD C8E9 4F3F 3834 3437 3D4C EEC8 BBB5

0060: B2B3 B8BF D35A 4138 3331 3339 435F CEBD B6B1 B1B4

0080: BBC9 7C48 3B34 3131 363D 4FDE

Tx Message 7: packet_length 100 channel_id 1 packet_id 0 process id 0x1

0000: 0000 4AC5 5F08 9131 0000 003C 66DE 66EB 67EE FE6E

0020: F7E7 6B68 E068 EE6A DF5C DF62 EDF1 6FF2 7A78 67DC

0040: 5EDF 62E7 64E6 66E0 7071 EA69 F86E E260 DE5D E665

0060: EB75 F0FB 6DE9 64E4 69E3 66EA 67E9 6DF9 F177 EC6E

0080: EB6E F876 F875 7D6E E966 E05D

Tx Message 8: packet_length 100 channel_id 2 packet_id 0 process id 0x1

0000: 0000 1CDB 3E48 3AD4 0000 003C C2B9 B3B1 B3B8 C2DC

0020: 523F 3733 3235 3C49 72CB BDB7 B4B5 B8BF CF67 483C

0040: 3836 373C 455C DAC6 BDB9 B9BB C0CC EE54 453E 3C3C

0060: 3F47 56F1 D1C7 C1BF C0C6 CEE1 6651 4A46 4648 4D59

0080: 70E3 D6CF CDCE D2D9 E67E 675E

Bootloader 1.8, Appn 3.1

Application firmware 3.1.8, Built by claux on Thu Jun 17 11:00:05 1999

VIC Interface Foreign Exchange Station 0/0, DSP instance (0x19543C0)

Singalling channel num 128 Signalling proxy 0x0 Signaling dsp 0x19543C0 tx outstanding 0, max tx outstanding 32 ptr 0x0, length 0x0, max length 0x0 dsp_number 0, Channel ID 1 received 0 packets, 0 bytes, 0 gaint packets

0 drops, 0 no buffers, 0 input errors 0 input overruns

650070 bytes output, 4976 frames output, 0 output errors, 0 output underrun

0 unaligned frames

VIC Interface Foreign Exchange Station 0/1, DSP instance (0x1954604)

Singalling channel num 129 Signalling proxy 0x0 Signaling dsp 0x1954604 tx outstanding 0, max tx outstanding 32 ptr 0x0, length 0x0, max length 0x0 dsp_number 0, Channel ID 2 received 0 packets, 0 bytes, 0 gaint packets

0 drops, 0 no buffers, 0 input errors 0 input overruns

393976 bytes output, 3982 frames output, 0 output errors, 0 output underrun

0 unaligned frames

Field descriptions are hardware-dependent and are meant for use by trained technical support.

Related Commands

Command show dial-peer voice show interface dspfarm show voice dsp show voice port

Description

Displays configuration information and call statistics for dial peers.

Displays hardware information including DRAM,

SRAM, and the revision-level information on the line card.

Displays the current status of all DSP voice channels.

Displays configuration information about a specific voice port.

402

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition show crm

show crm

To display the carrier call capacities statistics, use the show crm command in privileged EXEC mode.

show crm

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(11)T

Modification

This command was introduced.

Usage Guidelines

Both the show trunk groupcommandand the show crm command display values for the maximum number of calls. These values originate from different configuration procedures:

• In the show trunk group command, the Max Calls value originates from the max-calls command in the trunk group configuration.

• In the show crm command, Max calls indicates the maximum number of available channels after the carrier ID or trunk group label is assigned to an interface using the trunk-group (interface) command.

Examples

The following example illustrates the carrier call capacities statistics:

Router#

show crm

Carrier:1411

Max calls:4

Max Voice (in) :

Max Voice (out):

Max Data (in) :

Max Data (out) :

Trunk Group Label: 100

Max calls:6

Max Voice (in) :

Max Voice (out):

Max Data (in) :

Max Data (out) :

4

4

4

4

6

6

6

6

Cur Voice (in) :

Cur Voice (out):

Cur Data (in) :

Cur Data (out) :

Cur Voice (in) :

Cur Voice (out):

Cur Data (in) :

Cur Data (out) :

0

0

0

0

0

0

0

0

The table below describes the fields shown in this output, in alphabetical order.

Cisco IOS Voice Command Reference - S commands

403

show call history fax through show debug condition show crm

Table 60: show crm Field Descriptions

Field

Carrier

Cur Data (in)

Cur Data (out)

Cur Voice (in)

Cur Voice (out)

Max Calls

Max Data (in)

Max Data (out)

Max Voice (in)

Max Voice (out)

Trunk Group Label

Related Commands

Command

carrier-id (dial-peer) max-calls show trunk group trunk-group (interface) trunk-group-label (dial-peer)

404

Cisco IOS Voice Command Reference - S commands

Description

ID of the carrier that handles the calls.

Current number of incoming data calls that are handled by the carrier or trunk group.

Current number of outgoing data calls that are handled by the carrier or trunk group.

Current number of incoming voice calls that are handled by the carrier or trunk group.

Current number of outgoing voice calls that are handled by the carrier or trunk group.

Maximum number of calls that are handled by the carrier or trunk group.

Maximum number of incoming data calls that are handled by the carrier or trunk group.

Maximum number of outgoing data calls that are handled by the carrier or trunk group.

Maximum number of incoming voice calls that are handled by the carrier or trunk group.

Maximum number of outgoing voice calls that are handled by the carrier or trunk group.

Label of the trunk group that handles the calls.

Description

Specifies the carrier associated with VoIP calls.

Specifies the maximum number of calls handled by a trunk group.

Displays the configuration parameters for one or more trunk groups.

Assigns an interface to a trunk group.

Specifies the trunk group associated with VoIP calls.

show call history fax through show debug condition show crm

Cisco IOS Voice Command Reference - S commands

405

show call history fax through show debug condition show csm

show csm

To display the call switching module (CSM) statistics for a particular digital signal processor (DSP) channel, all DSP channels, or a specific modem or DSP channel, use the show csmcommand in privileged EXEC mode.

Cisco AS5300 Universal Access Server

show csm {call-rate [table]| callre-source| modem [slot/port| group modem-group-number]|

signaling-channel}

Cisco AS5400Series Router

show csm {call rate [table]| call-resource| modem [slotport| group modem-group-number]| signaling-channel|

voice slot/port}

Syntax Description call-rate table call-resource modem

slot / port

group

modem -group-number

signaling-channel voice

Displays the incoming and outgoing call switching rate.

(Optional) Displays the incoming and outgoing call switching rate in the form of numerical table.

Displays statistics about the CSM call resource.

Displays CSM call statistics for modems.

(Optional) Location (and thereby identity) of a specific modem.

(Optional) Displays modem group information.

(Optional) Location of a particular dial peer. Range:

1 to 32767.

Displays CSM signaling channel Information.

Displays CSM call statistics for DSP channels.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3 NA

Modification

This command was introduced.

406

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition show csm

Release

12.0(3)T

12.0(7)T

Modification

This command was modified. Port-specific values for the Cisco AS5300 were added.

This command was modified. Port-specific values for the Cisco AS5800 were added.

Usage Guidelines

This command shows the information related to CSM, which includes the DSP channel, the start time of the call, the end time of the call, and the channel on the controller used by the call.

Use the show csm modem command to display the CSM call statistics for a specific modem, for a group of modems, or for all modems. If a slot / port argument is specified, then CSM call statistics are displayed for the specified modem. If the modem-group-numberargument is specified, the CSM call statistics for all of the modems associated with that modem group are displayed. If no keyword is specified, CSM call statistics for all modems on the Cisco AS5300 universal access server are displayed.

Use the show csm voice command to display CSM statistics for a particular DSP channel. If the slot / dspm

/ dsp / dsp-channel or shelf / slot / portargumentis specified, the CSM call statistics for calls using the identified

DSP channel are displayed. If no argument is specified, all CSM call statistics for all DSP channels are displayed.

Examples

The following is sample output from the show csm command for the Cisco AS5300 universal access server:

Router#

show csm voice 2/4/4/0

slot 2, dspm 4, dsp 4, dsp channel 0, slot 2, port 56, tone, device_status(0x0002): VDEV_STATUS_ACTIVE_CALL.

csm_state(0x0406)=CSM_OC6_CONNECTED, csm_event_proc=0x600E2678, current call thru PRI line invalid_event_count=0, wdt_timeout_count=0 wdt_timestamp_started is not activated wait_for_dialing:False, wait_for_bchan:False pri_chnl=TDM_PRI_STREAM(s0, u0, c22), tdm_chnl=TDM_DSP_STREAM(s2, c27) dchan_idb_start_index=0, dchan_idb_index=0, call_id=0xA003, bchan_num=22 csm_event=CSM_EVENT_ISDN_CONNECTED, cause=0x0000 ring_no_answer=0, ic_failure=0, ic_complete=0 dial_failure=0, oc_failure=0, oc_complete=3 oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0 remote_link_disc=0, stat_busyout=0 oobp_failure=0 call_duration_started=00:06:53, call_duration_ended=00:00:00, total_call_duration=00:00:44

The calling party phone number = 408

The called party phone number = 5271086 total_free_rbs_timeslot = 0, total_busy_rbs_timeslot = 0, total_dynamic_busy_rbs_timeslot

= 0, total_static_busy_rbs_timeslot = 0, total_sw56_rbs_timeslot = 0, total_sw56_rbs_static_bo_ts = 0, total_free_isdn_channels = 21, total_busy_isdn_channels = 0,total_auto_busy_isdn_channels

= 0, min_free_device_threshold = 0

The table below describes the significant fields shown in the display.

Table 61: show csm voice Field Descriptions

Field

slot

Description

Slot where the VFC resides.

Cisco IOS Voice Command Reference - S commands

407

show csm

Field

dsp slot/port tone

show call history fax through show debug condition

Description

DSP through which this call is established.

Logical port number for the device. This is equivalent to the DSP channel number. The port number is derived as follows:

• (max_number_of_dsp_channels per dspm=12)

* the dspm # (0-based) +

• (max_number_of_dsp_channels per dsp=2) * the dsp # (0-based) + the dsp channel number

(0-based).

Which signaling tone is being used (DTMF, MF, R2).

This only applies to CAS calls. Possible values are as follows:

• mf

• dtmf

• r2-compelled

• r2-semi-compelled

• r2-non-compelled

408

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition

Field

device_status

show csm

Description

Status of the device. Possible values are as follows:

• VDEV_STATUS_UNLOCKED--Device is unlocked (meaning that it is available for new calls).

• VDEV_STATUS_ACTIVE_WDT--Device is allocated for a call and the watchdog timer is set to time the connection response from the central office.

• VDEV_STATUS_ACTIVE_CALL--Device is engaged in an active, connected call.

• VDEV_STATUS_BUSYOUT_REQ--Device is requested to busyout; does not apply to voice devices.

• VDEV_STATUS_BAD--Device is marked as bad and not usable for processing calls.

• VDEV_STATUS_BACK2BACK_TEST--Modem is performing back-to-back testing (for modem calls only).

• VDEV_STATUS_RESET--Modem needs to be reset (for modem only).

• VDEV_STATUS_DOWNLOAD_FILE--Modem is downloading a file (for modem only).

• VDEV_STATUS_DOWNLOAD_FAIL--Modem has failed during downloading a file (for modem only).

• VDEV_STATUS_SHUTDOWN--Modem is not powered up (for modem only).

• VDEV_STATUS_BUSY--Modem is busy (for modem only).

• VDEV_STATUS_DOWNLOAD_REQ--Modem is requesting connection (for modem only).

Cisco IOS Voice Command Reference - S commands

409

show csm

Field

csm_state

show call history fax through show debug condition

Description

410

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition

Field show csm

Description

CSM call state of the current call (PRI line) associated with this device. Possible values are as follows:

• CSM_IDLE_STATE--Device is idle.

• CSM_IC_STATE--A device has been assigned to an incoming call.

• CSM_IC1_COLLECT_ADDR_INFO--A device has been selected to perform ANI/DNIS address collection for this call. ANI/DNIS address information collection is in progress. The

ANI/DNIS is used to decide whether the call should be processed by a modem or a voice

DSP.

• CSM_IC2_RINGING--The device assigned to this incoming call has been told to get ready for the call.

• CSM_IC3_WAIT_FOR_SWITCH_OVER--A new device is selected to take over this incoming call from the device collecting the ANI/DNIS address information.

• CSM_IC4_WAIT_FOR_CARRIER--This call is waiting for the CONNECT message from the carrier.

• CSM_IC5_CONNECTED--This incoming call is connected to the central office.

• CSM_IC6_DISCONNECTING--This incoming call is waiting for a DISCONNECT message from the VTSP module to complete the disconnect process.

• CSM_OC_STATE --An outgoing call is initiated.

• CSM_OC1_REQUEST_DIGIT--The device is requesting the first digit for the dial-out number.

• CSM_OC2_COLLECT_1ST_DIGIT--The first digit for the dial-out number has been collected.

• CSM_OC3_COLLECT_ALL_DIGIT--All the digits for the dial-out number have been collected.

• CSM_OC4_DIALING--This call is waiting for a dsx0 (B channel) to be available for dialing out.

• CSM_OC5_WAIT_FOR_CARRIER--This

Cisco IOS Voice Command Reference - S commands

411

show csm

Field

csm_state: invalid_event_count wdt_timeout_count wdt_timestamp_started wait_for_dialing wait_for_bchan pri_chnl tdm_chnl dchan_idb_start_index dchan_idb_index

412

Cisco IOS Voice Command Reference - S commands show call history fax through show debug condition

Description

(outgoing) call is waiting for the central office to connect.

• CSM_OC6_CONNECTED--This (outgoing) call is connected.

• CSM_OC7_BUSY_ERROR--A busy tone has been sent to the device (for VoIP call, no busy tone is sent; just a DISCONNECT

INDICATION message is sent to the VTSP module), and this call is waiting for a

DISCONNECT message from the VTSP module

(or ONHOOK message from the modem) to complete the disconnect process.

• CSM_OC8_DISCONNECTING--The central office has disconnected this (outgoing) call, and the call is waiting for a DISCONNECT message from the VTSP module to complete the disconnect process.

Number of invalid events received by the CSM state machine.

Number of times the watchdog timer is activated for this call.

Whether the watchdog timer is activated for this call.

Whether this (outgoing) call is waiting for a free digit collector to become available to dial out the outgoing digits.

Whether this (outgoing) call is waiting for a B channel to send the call out on.

Which type of TDM stream is used for the PRI connection. For PRI and CAS calls, it is always

TDM_PRI_STREAM.

Which type of TDM stream is used for the connection to the device used to process this call. In the case of a VoIP call, this is always set to

TDM_DSP_STREAM.

First index to use when searching for the next IDB of a free D channel.

Index of the currently available IDB of a free D channel.

show call history fax through show debug condition

Field

csm_event cause ring_no_answer ic_failure ic_complete dial_failure oc_failure oc_complete oc_busy oc_no_dial_tone oc_dial_timeout call_duration_started call_duration_ended total_call_duration

The calling party phone number

The called party phone number total_free_rbs_time slot total_busy_rbs_time slot total_dynamic_busy_rbs_time slot

show csm

Description

Event just passed to the CSM state machine.

Event cause.

Number of times a call failed because there was no response.

Number of failed incoming calls.

Number of successful incoming calls.

Number of times a connection failed because there was no dial tone.

Number of failed outgoing calls.

Number of successful outgoing calls.

Number of outgoing calls whose connection failed because there was a busy signal.

Number of outgoing calls whose connection failed because there was no dial tone.

Number of outgoing calls whose connection failed because the timeout value was exceeded.

Start of this call.

End of this call.

Duration of this call.

Calling party number as given to CSM by ISDN.

Called party number as given to CSM by ISDN.

Total number of free RBS (CAS) time slots available for the whole system.

Total number of RBS (CAS) time slots that have been busied-out. This includes both dynamically and statically busied out RBS time slots.

Total number of RBS (CAS) time slots that have been dynamically busied out.

Cisco IOS Voice Command Reference - S commands

413

show csm

Field

total_static_busy_rbs_time slot total_free_isdn_channels total_busy_isdn_channels total_auto_busy_isdn_channels

Related Commands

Command show call active voice show call history voice show num-exp show voice port show call history fax through show debug condition

Description

Total number of RBS (CAS) time slots that have been statically busied out (that is, they are busied out using the CLI command).

Total number of free ISDN channels.

Total number of busy ISDN channels.

Total number of ISDN channels that are automatically busied out.

Description

Displays the contents of the active call table.

Displays the contents of the call history table.

Displays how number expansions are configured.

Displays configuration information about a specific voice port.

414

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition show csm call

show csm call

To view the call switching module (CSM) call statistics, use the show csm call command in privileged EXEC mode

show csm call {failed| rate| total}

Syntax Description failed rate total

CSM call fail/reject rate for the last 60 seconds, 60 minutes, and 72 hours.

CSM call rate for the last 60 seconds, 60 minutes, and

72 hours.

Total number of CSM calls for the last 60 seconds,

60 minutes, and 72 hours.

Command Default

No default behavior or values.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(2)T

Modification

This command was introduced on the Cisco AS5850.

Usage Guidelines

Use this command to understand CSM call volume.

Examples

The following examples show the CSM call statistics for the last 60 seconds:

Router#

show csm call rate

6

5

4

3

2

15

14

13

12

11

10

9

8

7

Cisco IOS Voice Command Reference - S commands

415

show csm call show call history fax through show debug condition

1

0....5....1....1....2....2....3....3....4....4....5....5....

0 5 0 5 0 5 0 5 0 5

CSM call switching rate per second (last 60 seconds)

# = calls entering the module per second

Router#

show csm call failed

5

4

3

2

1

10

9

8

7

6

15

14

13

12

11

0....5....1....1....2....2....3....3....4....4....5....5....

0 5 0 5 0 5 0 5 0 5

CSM call fail/reject rate per second (last 60 seconds)

# = calls failing per second

Router#

sh csm call total

1344

1244

1144

1044

944

844

744

644

544

444

344

244

144

44

0....5....1....1....2....2....3....3....4....4....5....5....

0 5 0 5 0 5 0 5 0 5

CSM total calls (last 60 seconds)

# = number of calls

Field descriptions should be self-explanatory.

416

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition show cube debug category codes

show cube debug category codes

To display Cisco Unified Border Element debug category code information, use the show cube debug category

codes command in user EXEC or privileged EXEC mode.

show cube debug category codes

Syntax Description

This command has no arguments or keywords.

Command Modes

User EXEC (>)

Privileged EXEC (#)

Command History

Release

15.3(3)M

Modification

This command was introduced.

Examples

Device#

show cube debug category codes

|-----------------------------------------------

| show cube debug category codes values.

|-----------------------------------------------

| Indx | Debug Name | Value

|-----------------------------------------------

| 01 | SDP Debugs

| 02 | Audio Debugs

| 03 | Video Debugs

| 04 | Fax Debugs

|

|

|

|

1

2

4

8

| 05 | SRTP Debugs

| 06 | DTMF Debugs

|

|

16

32

| 07 | SIP Profiles Debugs | 64

| 08 | SDP Passthrough Deb | 128

| 09 | Transcoder Debugs | 256

| 10 | SIP Transport Debugs | 512

| 11 | Parse Debugs | 1024

| 12 | Config Debugs

| 13 | Control Debugs

| 2048

| 4096

| 14 | Miscellaneous Debugs | 8192

| 15 | Supp Service Debugs | 16384

| 16 | Misc Features Debugs| 32768

| 17 | SIP Line-side Debugs | 65536

| 18 | CAC Debugs | 131072

| 19 | Registration Debugs | 262144

|-----------------------------------------------

Cisco IOS Voice Command Reference - S commands

417

show call history fax through show debug condition show cube status

show cube status

To display the Cisco Unified Border Element (Cisco UBE) status, the software version, the license capacity, the image version, and the platform name of the device, use the show cube status command in user EXEC or privileged EXEC mode.

show cube status

Syntax Description

This command has no arguments or keywords.

Command Default

Cisco UBE status is not displayed.

Command Modes

User EXEC (>)

Privileged EXEC (#)

Command History

Release

15.2(1)T

15.1(3)S1

Modification

This command was introduced.

This command was modified.

The output was modified to have only token characters (an alphanumeric character, hyphen [-], dot [.], exclamation mark [!], percent [%], asterisk [*], underscore [_], plus sign [+], grave [`], apostrophe ['], or a tilde [~]) in server and user-agent Session

Initiation Protocol (SIP) headers. The nontoken characters present in the image name is replaced by a dot[.].

Usage Guidelines

The display of Cisco UBE status-related information is supported by the implementation of the

CISCO-UBE-MIB. This MIB also provides Simple Network Management Protocol (SNMP) support for the

Cisco UBE status:

The Cisco UBE status display is enabled only if the mode border-element command is configured with call license capacity. The show cube status command displays the following message if the license capacity is not configured.

Cisco Unified Border Element (CUBE) application is not enabled

Examples

The following example configures the mode border-element command with call license capacity and enables the display of Cisco UBE status on the Cisco 3845 router:

Device(config)#

voice service voip

Device(conf-voi-serv)#

mode border-element license capacity 200

After saving the configuration and reloading the device:

Device>

show cube status

418

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition show cube status

CUBE-Version : 11.0.0

SW-Version : 15.5(2)T, Platform 3845

HA-Type : none

Licensed-Capacity : 200

In Cisco IOS Release 15.1(3)S1 and later releases, the output is as follows:

Device>

show cube status

CUBE-Version : 8.8

SW-Version : 15.2.1.T, Platform 3845

HA-Type : none

Licensed-Capacity : 200

The table below describes the fields shown in the display.

Table 62: show cube status Field Descriptions

Field

CUBE-Version

SW-Version

HA-Type

Description

Version of the Cisco UBE application running on the device.

Image version and platform name of the device running the Cisco UBE application.

This matches the image version and platform name returned by the show version command.

The type of High Availability (HA) feature configured and running on the device.

The following HA types are supported:

• none: Cisco UBE does not support HA.

• cold-standby-chassis-to-chassis: Device-to-device cold standby support.

• hot-standby-chassis-to-chassis: Device-to-device hot standby support.

Licensed-Capacity Number of SIP call legs that Cisco UBE is licensed to use. The range is from 0 to

999999. This number matches the number of licenses configured using the mode

border-element license capacity command.

Note

The number of SIP call legs that Cisco UBE can use is platform-dependent and is not affected by the specified value for the capacity keyword in

Cisco IOS Release 15.2(1)T.

Related Commands

Command mode border-element

Description

Enables the set of commands used in the border-element configuration on the Cisco 2900 and

Cisco 3900 series platforms.

Cisco IOS Voice Command Reference - S commands

419

show call history fax through show debug condition show debug condition

show debug condition

To display the debugging filters that have been enabled for VoiceXML applications, ATM-enabled interfaces, or Frame Relay interfaces, use the show debug condition command in privileged EXEC mode.

show debug condition

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(11)T

12.0(28)S

12.2(25)S

12.2(27)SBC

12.2(28)SB

12.4(9)T

Modification

This command was introduced on the Cisco 3640, Cisco 3660, Cisco AS5300,

Cisco AS5350, and Cisco AS5400.

This command was integrated into Cisco IOS Release 12.0(28)S and was enhanced to include debugging for ATM-enabled and Frame Relay-enabled interfaces.

This command was integrated into Cisco IOS Release 12.2(25)S.

This command was integrated into Cisco IOS Release 12.2(27)SBC.

This command was integrated into Cisco IOS Release 12.2(28)SB.

This command was enhanced to include debugging for ATM-enabled and

Frame Relay-enabled interfaces.

Usage Guidelines

This command displays the debugging filter conditions that have been set for VoiceXML applications by using the debug condition application voice command.

Examples

The following is sample output from this command when it is used with the VoiceXML application:

Router#

show debug condition

Condition 1: application voice vmail (1 flags triggered)

Flags: vmail

Condition 2: application voice myapp1 (1 flags triggered)

Flags: myapp1

The following is sample output from this command when an ATM interface is being debugged:

Router#

show debug condition

Condition 1: atm-vc 0/56784 AT2/0 (0 flags triggered)

Condition 2: atm-vc 255/45546 AT2/0 (0 flags triggered)

Condition 3: atm-vc 0/266 AT6/0 (1 flags triggered)

420

Cisco IOS Voice Command Reference - S commands

show call history fax through show debug condition show debug condition

The table below describes the significant fields shown in the display.

Table 63: show debug condition Field Descriptions

Field

Condition 1

Flags at2/0 atm-vc 0/56784

Description

Sequential number identifying the filter condition that was set for the specified command.

Name of the voice application for which the condition was set.

Interface number of the ATM interface that has the debug condition applied.

Virtual channel identifier (VCI). Alternatively, virtual path identifier/virtual channel identifier (VCI/VPI) pair.

Related Commands

Command debug condition application voice debug http client debug vxml

Description

Filters out debugging messages for all VoiceXML applications except the specified application.

Displays debugging messages for the HTTP client.

Displays debugging messages for VoiceXML features.

Cisco IOS Voice Command Reference - S commands

421

show debug condition show call history fax through show debug condition

422

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

show dial-peer, page 425

show dial-peer video, page 426

show dial-peer voice, page 428

show dialplan dialpeer, page 439

show dialplan incall, page 446

show dialplan incall uri, page 449

show dialplan in-carrier, page 456

show dialplan in-trunk-group-label, page 458

show dialplan number, page 460

show dialplan uri, page 463

show dn-numbers, page 466

show dspfarm, page 468

show dspfarm profile, page 475

show dsp-group, page 479

show echo-cancel, page 481

show event-manager consumers, page 483

show frame-relay vofr, page 485

show gatekeeper calls, page 488

show gatekeeper circuits, page 491

show gatekeeper cluster, page 493

show gatekeeper endpoint circuits, page 494

show gatekeeper endpoints, page 497

show gatekeeper gw-type-prefix, page 501

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423

show gatekeeper performance statistics, page 504

show gatekeeper servers, page 508

show gatekeeper status, page 511

show gatekeeper status cluster, page 513

show gatekeeper zone cluster, page 514

show gatekeeper zone prefix, page 515

show gatekeeper zone status, page 517

show dial-peer through show gatekeeper zone prefix

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show dial-peer through show gatekeeper zone prefix show dial-peer

show dial-peer

To display the dial plan mapping table for protocol peers, use the show dial-peercommand in privileged

EXEC mode.

show dial-peer {carrier| cor| trunk-group-label}

Syntax Description carrier cor trunk-group-label

Displays carrier ID configuration details of the peer protocol.

Displays restriction settings class details.

Displays trunk group label configuration details.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(17)SX

12.4(22)T

Modification

This command was introduced.

This command was modified in a release earlier than Cisco IOS Release

12.4(22)T. The carrier and trunk-group-labelkeywords were added.

Usage Guidelines

Use this command to display the dial plan mapping table for protocol peers along with the available keywords.

Examples

The following sample output from the show dial-peer command displays restriction settings class details.

The fields are self-explanatory.

Router#

show dial-peer cor

Class of Restriction name: class1

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425

show dial-peer through show gatekeeper zone prefix show dial-peer video

show dial-peer video

To display configuration information for video dial peers, use the show dial-peer videocommand in privileged

EXEC mode.

show dial-peer video [ number ] [summary]

Syntax Description

number

summary

(Optional) A specific video dial peer. Output displays information about that dial peer.

(Optional) Output displays a one-line summary of each video dial peer.

Command Default

If both the name argument and summary keyword are omitted, command output displays detailed information about all video dial peers.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(5)XK

12.0(7)T

Modification

This command was introduced on the Cisco MC3810.

This command was integrated into Cisco IOS Release 12.0(7)T.

Usage Guidelines

Use this command to display the configuration for all video dial peers configured for a router. To show configuration information for only one specific dial peer, use the number argument to identify the dial peer.

Examples

The following sample output displays detailed information about all configured video dial peers:

Router#

show dial-peer video

Video Dial-Peer 1 type = videocodec, destination-pattern = 111 port signal = 1/0, port media = Serial1 nsap = 47.0091810000000050E201B101.00107B09C6F2.C8

Video Dial-Peer 2 type = videoatm, destination-pattern = 222 session-target = ATM0 svc nsap 47.0091810000000050E201B101.00E01E92ADC2.C8

Video Dial-Peer 3 type = videoatm, destination-pattern = 333 session-target = ATM0 pvc 70/70

The table below describes the significant fields shown in the output.

426

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Table 64: show dial-peer video Field Descriptions

Field

NSAP

Description

Network service access point (NSAP) address

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427

show dial-peer through show gatekeeper zone prefix show dial-peer voice

show dial-peer voice

To display information for voice dial peers, use the show dial-peer voicecommand in user EXEC or privileged

EXEC mode.

show dial-peer voice [number| busy-trigger-counter| summary| voip system]

Syntax Description

number

busy-trigger-counter summary voip system

(Optional) A specific voice dial peer. The output displays detailed information about that dial peer.

(Optional) Displays the busy trigger call count on the

VoIP dial peer.

(Optional) Displays a short summary of each voice dial peer.

(Optional) Displays information about the VoIP dial peer.

Command Default

If both the number argument and summary keyword are omitted, the output displays detailed information about all voice dial peers.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

11.3(1)T

11.3(1)MA

12.0(3)XG

12.0(4)T

12.1(3)T

12.2(2)XB

12.2(4)T

Modification

This command was introduced.

This command was modified. The summary keyword was added for the Cisco

MC3810.

This command was implemented for Voice over Frame Relay (VoFR) on the

Cisco 2600 series and Cisco 3600 series.

This command was implemented for VoFR on the Cisco 7200 series.

This command was implemented for modem pass-through over VoIP on the Cisco

AS5300.

This command was modified to support VoiceXML applications.

This command was implemented on the Cisco 1750.

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show dial-peer through show gatekeeper zone prefix show dial-peer voice

Release

12.2(8)T

12.2(2)XN

12.2(11)T

12.4(22)T

15.0(1)XA

15.1(1)T

15.1(3)T

Modification

This command was implemented on the Cisco 1751, Cisco 2600 series, Cisco

3600 series, Cisco 3725, and Cisco 3745.

This command was modified. Support for enhanced Media Gateway Control

Protocol (MGCP) voice gateway interoperability was added to Cisco CallManager

3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.

This command was integrated into Cisco IOS Release 12.2(11)T and Cisco

CallManager 3.2 and implemented on the Cisco IAD2420. The command was enhanced to display configuration information for bandwidth, video codec, and rtp payload-type for H.263+ and H.264 video codec.

This command was modified. This command was enhanced to display the current configuration state of the history-info header. Command output was updated to show IPv6 information.

This command was modified. The output was enhanced to show the logical partitioning class of restriction (LPCOR) policy for outgoing calls.

This command was integrated into Cisco IOS Release 15.1(1)T.

This command was modified. The output was enhanced to display information about the bind at the dial-peer level and to display the connection status of Foreign

Exchange Office (FXO) ports.

Usage Guidelines

Use this command to display the configuration for all VoIP and POTS dial peers configured for a gateway.

To display configuration information for only one specific dial peer, use the number argument. To display summary information for all dial peers, use the summary keyword.

Note

The recommended command to verify the QoS settings that the signaling and media packets will be marked with when RSVP is not configured for call signaling on the Cisco UBE is the show dial-peer voice command.

Examples

The following is sample output from the show dial-peer voicecommand for a POTS dial peer:

Router#

show dial-peer voice 100

VoiceEncapPeer3201 peer type = voice, information type = video, description = `', tag = 3201, destination-pattern = `86001', answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent

CLID Override RDNIS = disabled, source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown'

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show dial-peer through show gatekeeper zone prefix show dial-peer voice

group = 3201, Admin state is up, Operation state is up,

Outbound state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled,

URI classes:

Destination = huntstop = disabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4 type = pots, prefix = `', forward-digits 4 session-target = `', voice-port = `2/0:23', direct-inward-dial = enabled, digit_strip = enabled, register E.164 number with H323 GK and/or SIP Registrar = TRUE fax rate = system, payload size = 20 bytes supported-language = '' preemption level = `routine' bandwidth: maximum = 384 KBits/sec, minimum = 64 KBits/sec voice class called-number: inbound = `', outbound = `1'

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

The following is sample output from this command for a VoIP dial peer:

Router#

show dial-peer voice 101

VoiceOverIpPeer101 peer type = voice, system default peer = FALSE, information type = voice, description = `', tag = 1234, destination-pattern = `', voice reg type = 0, corresponding tag = 0, allow watch = FALSE answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent

CLID Override RDNIS = disabled, rtp-ssrc mux = system source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 1234, Admin state is up, Operation state is down, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, modem transport = system,

URI classes:

Incoming (Request) =

Incoming (Via) =

Incoming (To) =

Incoming (From) =

Destination = huntstop = disabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both

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incoming COR list:maximum capability outgoing COR list:minimum requirement outgoing LPCOR:

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4 mailbox selection policy: none type = voip, session-target = `', technology prefix: settle-call = disabled ip media DSCP = ef, ip media rsvp-pass DSCP = ef ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31, ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41 ip video rsvp-fail DSCP = af41, ip defending Priority = 0, ip preemption priority = 0 ip policy locator voice: ip policy locator video:

UDP checksum = disabled, session-protocol = sipv2, session-transport = system, req-qos = best-effort, acc-qos = best-effort, req-qos video = best-effort, acc-qos video = best-effort, req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0, req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,

RTP dynamic payload type values: NTE = 101

Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122

CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,

A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, iSAC=124 lmr_tone=0, nte_tone=0 h263+=118, h264=119

G726r16 using static payload

G726r24 using static payload

RTP comfort noise payload type = 19 fax rate = voice, payload size = 20 bytes fax protocol = system fax-relay ecm enable

Fax Relay ans enabled

Fax Relay SG3-to-G3 Enabled (by system configuration) fax NSF = 0xAD0051 (default) codec = g729r8, payload size = 20 bytes, video codec = None voice class codec = `' voice class sip session refresh system voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30 voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30 voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30 voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30 text relay = disabled

Media Setting = forking (disabled) flow-through (global)

Expect factor = 10, Icpif = 20,

Playout Mode is set to adaptive,

Initial 60 ms, Max 1000 ms

Playout-delay Minimum mode is set to default, value 40 ms

Fax nominal 300 ms

Max Redirects = 1, signaling-type = cas,

VAD = enabled, Poor QOV Trap = disabled,

Source Interface = NONE voice class sip url = system, voice class sip tel-config url = system, voice class sip rel1xx = system, voice class sip anat = system, voice class sip outbound-proxy = "system", voice class sip associate registered-number = system, voice class sip asserted-id system, voice class sip privacy system voice class sip e911 = system, voice class sip history-info = system, voice class sip reset timer expires 183 = system, voice class sip pass-thru headers = system, voice class sip pass-thru content unsupp = system, voice class sip pass-thru content sdp = system,

Cisco IOS Voice Command Reference - S commands

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show dial-peer through show gatekeeper zone prefix show dial-peer voice

voice class sip copy-list = system, voice class sip g729 annexb-all = system, voice class sip early-offer forced = system, voice class sip negotiate cisco = system, voice class sip block 180 = system, voice class sip block 183 = system, voice class sip block 181 = system, voice class sip preloaded-route = system, voice class sip random-contact = system, voice class sip random-request-uri validate = system, voice class sip call-route p-called-party-id = system, voice class sip call-route history-info = system, voice class sip privacy-policy send-always = system, voice class sip privacy-policy passthru = system, voice class sip privacy-policy strip history-info = system, voice class sip privacy-policy strip diversion = system, voice class sip map resp-code 181 = system, voice class sip bind control = enabled, 9.42.28.29, voice class sip bind media = enabled, 9.42.28.29, voice class sip bandwidth audio = system, voice class sip bandwidth video = system, voice class sip encap clear-channel = system, voice class sip error-code-override options-keepalive failure = system, voice class sip calltype-video = false voice class sip registration passthrough = System voice class sip authenticate redirecting-number = system, redirect ip2ip = disabled local peer = false probe disabled,

Secure RTP: system (use the global setting) voice class perm tag = `'

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

Last Disconnect Time = 0.

When there is no Dial-peer level bind voice class sip bind control = system, voice class sip bind media = system,

The following is sample output from the show dial-peer voice summary command that shows connected

FXO port 0/2/0 (the last entry) has OUT STAT set to "up," which indicates that the POTS dial peer can be used for an outgoing call. If this port is disconnected, the status changes in the output so that the OUT STAT field reports "down," and the POTS dial peer cannot be used for an outgoing call.

Note

Beginning in Cisco IOS Release 15.1(3)T, there is improved status monitoring of FXO ports--any time an FXO port is connected or disconnected, a message is displayed to indicate the status change. For example, the following message is displayed to report that a cable has been connected, and the status is changed to "up" for FXO port 0/2/0: 000118: Jul 14 18:06:05.122 EST: %LINK-3-UPDOWN: Interface

Foreign Exchange Office 0/2/0, changed state to operational status up due to cable reconnection

Router#

show dial-peer voice summary

dial-peer hunt 0

TAG

KEEPALIVE

AD

TYPE MIN OPER PREFIX

39275- voip up up

82

DEST-PATTERN

.T

8880 pots up up

8881 pots up up

8882 pots up up

8883 pots up up

8884 pots up up

8880

8881

8882

8883

8884

PRE PASS

FER THRU SESS-TARGET

OUT

STAT PORT

0 syst ipv4:172.18.108.26

0

0

0

0

0 up up

2/0/0

2/0/1 up 2/0/2 up 2/0/3 up 2/0/4

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8885 pots up up

8886 pots up up

8887 pots up up

88888- pots up up

888

65033- pots up up

52

8885

8886

8887

6503352

0

0

0

0

0 up 2/0/5 up 2/0/6 up 2/0/7 down 0/3/0:23 up 0/2/0

The table below describes the significant fields shown in the displays, in alphabetical order.

Table 65: show dial-peer voice Field Descriptions

Field

Accepted Calls acc-qos

Admin state answer-address bandwidth maximum/minimum

Charged Units

CLID Restriction

CLID Network Number

CLID Second Number sent

CLID Override RDNIS codec

Connect Time connections/maximum

Destination

Description

Number of calls accepted from this peer since system startup.

Lowest acceptable quality of service configured for calls for this peer.

Administrative state of this peer.

Answer address configured for this dial peer.

The maximum and minimum bandwidth, in Kb/s.

Total number of charging units that have applied to this peer since system startup, in hundredths of a second.

Indicates if Calling Line ID (CLID) restriction is enabled.

Displays the network number sent as CLID, if configured.

Displays whether a second calling number is stripped from the call setup.

Indicates whether the CLID is overridden by the redirecting number.

Default voice codec rate of speech.

Accumulated connect time to the peer since system startup for both incoming and outgoing calls, in hundredths of a second.

Indicates the maximum number of call connections per peer.

Indicates the voice class that is used to match the destination URL.

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433

show dial-peer voice

Field

destination-pattern digit_strip direct-inward-dial disconnect-cause dnis-map

DTMF Relay

Expect factor

Failed Calls fax rate forward-digits group huntstop

Icpif in bound application associated incall-number incoming call blocking incoming called-number incoming COR list

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Cisco IOS Voice Command Reference - S commands show dial-peer through show gatekeeper zone prefix

Description

Destination pattern (telephone number) for this peer.

Indicates if digit stripping is enabled.

Indicates if direct inward dial is enabled.

Indicates the disconnect cause code to be used when an incoming call is blocked.

Name of the dialed-number identification service

(DNIS) map.

Indicates if dual-tone multifrequency (DTMF) relay is enabled.

User-requested expectation factor of voice quality for calls through this peer.

Number of failed call attempts to this peer since system startup.

Fax transmission rate configured for this peer.

Indicates the destination digits to be forwarded of this peer.

Group number associated with this peer.

Indicates whether dial-peer hunting is turned on, by the huntstop command, for this dial peer.

Configured Impairment/Calculated Planning

Impairment Factor (ICPIF) value for calls sent by a dial peer.

Interactive voice response (IVR) application that is configured to handle inbound calls to this dial peer.

Full E.164 telephone number to be used to identify the dial peer.

Indicates the incoming call blocking setup of this peer.

Indicates the incoming called number if it has been set.

Indicates the level of Class of Restrictions for incoming calls of this peer.

show dial-peer through show gatekeeper zone prefix

Field

Incomplete Calls information type

Last Disconnect Cause

Last Disconnect Text

Last Setup Time

Modem passthrough numbering Type

Operation state outgoing COR list outgoing LPCOR out bound application associated

Outbound state payload size payload type peer type permission

Poor QOV Trap preemption level

show dial-peer voice

Description

Indicates the number of outgoing disconnected calls with the user busy (17), no user response (18), or no answer (19) cause code.

Information type for this call (voice, fax, video).

Encoded network cause associated with the last call.

This value is updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.

ASCII text describing the reason for the last call termination.

Value of the system uptime when the last call to this peer was started.

Modem pass-through signaling method is named signaling event (NSE).

Indicates the numbering type for a peer call leg.

Operational state of this peer.

Indicates the level of Class of Restrictions for outgoing calls of this peer.

Setting of the lpcor outgoing command.

The voice application that is configured to handle outbound calls from this dial peer. Outbound calls are handed off to the named application.

Indicates the current outbound status of a POTS peer.

Indicates the size (in bytes) of the payload of the fax rate or codec setup.

NSE payload type.

Dial peer type (voice, data).

Configured permission level for this peer.

Indicates if poor quality of voice trap messages is enabled.

Indicates the call preemption level of this peer.

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show dial-peer through show gatekeeper zone prefix show dial-peer voice

Field

prefix

Redundancy

Refused Calls

Description

Indicates dialed digits prefix of this peer.

Packet redundancy (RFC 2198) for modem traffic.

Number of calls from this peer refused since system startup.

Indicates the "register e.164" option of this peer.

register E.164 number with H.323 GK and/or SIP

Registrar req-qos session-target session-protocol

Configured requested quality of service for calls for this dial peer.

Session target of this peer.

Session protocol to be used for Internet calls between local and remote routers through the IP backbone.

source carrier-id source trunk-group label

Successful Calls supported-language tag

Indicates the source carrier ID of this peer that will be used to match the source carrier ID of an incoming call.

Indicates the source trunk group label of this peer that can be used to match the source trunk group label of an incoming call.

Number of completed calls to this peer.

Indicates the list of supported languages of this peer.

Unique dial peer ID number.

target carrier-id target-trunkgroup-label

Indicates the target carrier ID of this peer that will be used to match the target carrier ID for an outgoing call.

Indicates the target trunk group label of this peer that can be used to match the target trunk group label of an outgoing call.

Time elapsed since last clearing of voice call statistics Elapsed time between the current time and the time when theclear dial-peer voice command was executed.

Translation profile (Incoming) Indicates the translation profile for incoming calls.

Translation profile (Outgoing) translation-profile

Indicates the translation profile for outgoing calls.

Indicates the number translation profile of this peer.

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Field

type

VAD voice class called-number inbound/outbound voice class sip history-info voice class sip bind voice-port

Description

Indicates the peer encapsulation type (pots, voip, vofr, voatm or mmoip).

Whether voice activation detection (VAD) is enabled for this dial peer.

Indicates the voice-class called number inbound or outbound setup of this peer.

Indicates the configuration state of the history-info header. If the history-info header is not configured for the dial peer, this field is set to system. If the history-info header is enabled on this dial peer, this field is set to enable. If the history-info header is disabled on this dial peer, this field is set to disable.

Indicates the configuration state of the bind address.

If the bind is configured for the global, this field is sent to system. If the bind address is enabled on this dial peer, this field is set to enabled.

Indicates the voice interface setting of this POTS peer.

The following is sample output from this command with the summary keyword:

Router#

show dial-peer voice summary

dial-peer hunt 0

TAG TYPE ADMIN OPER PREFIX DEST-PATTERN

100 pots up up

101 voip up

102 voip up

99 voip up

33 pots up up up down down

5550112

5550134

PASS

PREF THRU SESS-TARGET

0

0 syst ipv4:10.10.1.1

0 syst ipv4:10.10.1.1

0 syst

0

The table below describes the significant fields shown in the display.

PORT

Table 66: show dial-peer voice summary Field Descriptions

Field

dial-peer hunt

TAG

TYPE

ADMIN

Description

Hunt group selection order that is defined for the dial peer by the dial-peer hunt command.

Unique identifier assigned to the dial peer when it was created.

Type of dial peer (mmoip, pots, voatm, vofr, or voip).

Whether the administrative state is up or down.

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show dial-peer voice

Field

OPER

PREFIX

DEST-PATTERN

PREF

PASS THRU

SESS-TARGET

PORT

Related Commands

Command show call active voice show call history voice show dialplan incall number show dialplan number show num-exp show voice port show dial-peer through show gatekeeper zone prefix

Description

Whether the operational state is up or down.

Prefix that is configured in the dial peer by the prefix command.

Destination pattern that is configured in the dial peer by the destination-pattern command.

Hunt group preference that is configured in the dial peer by the preference command.

Modem pass-through method that is configured in the dial peer by the modem passthrough command.

Destination that is configured in the dial peer by the

session target command.

Router voice port that is configured for the dial peer.

Valid only for POTS dial peers.

Description

Displays the VoIP active call table.

Displays the VoIP call history table.

Displays which POTS dial peer is matched for a specific calling number or voice port.

Displays which dial peer is reached when a specific telephone number is dialed.

Displays how the number expansions are configured in VoIP.

Displays configuration information about a specific voice port.

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show dial-peer through show gatekeeper zone prefix show dialplan dialpeer

show dialplan dialpeer

To display the outbound dial peers that are matched to an incoming dial peer based on the class of restriction

(COR) criteria and the dialed number, use the show dialplan dialpeer command in privileged EXEC mode.

show dialplan dialpeer incoming-dialpeer-tag number number [timeout]

Syntax Description

incoming-dialpeer-tag

number

timeout

The dial peer COR identifier used to determine the matching outbound dial peer.

The dialed number used in conjunction with the COR identifier to determine the matching outbound dial peer.

(Optional) Allows matching for variable-length destination patterns.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(3)T

12.2(11)T

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on Cisco AS5800 access servers.

This command was implemented on the Cisco 1751 and Cisco 3700 series routers and on Cisco AS5300 access servers.

Usage Guidelines

Use this command as a troubleshooting tool to determine which outbound dial peer is matched for an incoming call, based on the COR criteria and dialed number specified in the command line. Use the timeout keyword to enable matching variable-length destination patters associated with dial peers. This can increase your chances of finding a match for the dial peer number you specify.

Note

For actual voice calls coming into the router, the incoming corlist of a specified inbound dial peer and the outgoing called number will be used to match the outbound dial peer.

Cisco IOS Voice Command Reference - S commands

439

show dial-peer through show gatekeeper zone prefix show dialplan dialpeer

Examples

The following sample output shows an incoming call with a dialed number of 19001111 and meeting the COR criteria as part of dial peer 300 with incoming COR-list has been matched to an outbound dial peer with IP address 1.8.50.7:

Router#

show dialplan dialpeer 300 number 1900111

VoiceOverIpPeer900 information type = voice, description = `', tag = 900, destination-pattern = `1900', answer-address = `', preference=0, numbering Type = `unknown' group = 900, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, modem passthrough = system, huntstop = disabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:to900 type = voip, session-target = `ipv4:1.8.50.7', technology prefix: settle-call = disabled

...

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

Matched: 19001111 Digits: 4

Target: ipv4:1.8.50.7

The table below describes the significant fields shown in the display.

Table 67: show dialplan command Field Descriptions

Field

Macro Exp.

VoiceEncapPeer

VoiceOverIpPeer peer type information type description tag destination-pattern

Description

Expected destination pattern for this dial peer.

Dial peer associated with the calling number entered.

Dial peer associated with the calling number entered.

Type of this dial peer (voice or data).

Information type for this dial peer (voice or data).

Any additional information for this dial peer entered using the description dial peer command.

Unique number identifying the dial peer.

Destination pattern (telephone number) configured for this dial peer.

440

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Field

answer-address preference

CLID restriction

CLID Network Number

CLID Second Number sent source carrier-id source trunk-group-label numbering Type group

Admin state

Operation state incoming called-number connections/maximum

DTMF Relay

URI classes: Incoming (Request)

URI classes: Incoming (To)

URI classes: Incoming (From)

URI classes: Destination

show dialplan dialpeer

Description

Answer address (calling number) configured for this dial peer.

Hunt group preference order set for this dial peer.

Indicates the Caller ID restriction (if any) configured for this dial peer.

Indicates the originating network of the Caller ID source.

Indicates the digits in the second number (if any) forwarded for this dial peer.

VoIP or POTS source carrier identifier.

VoIP or POTS source trunk group identifier.

Identifies the numbering scheme employed for this dial peer.

Dial peer group in which this dial peer is a member.

Administrative state of this dial peer.

Operational state of this dial peer.

Called number (DNIS) configured for this dial peer.

Number of actual and maximum allowable connections associated with this dial peer.

Whether the dtmf-relay command is enabled or disabled for this dial peer.

URI voice class used for matching dial peer to

Request-URI in an incoming SIP Invite message.

URI voice class used for matching dial peer to the To header in an incoming SIP Invite message.

URI voice class used for matching dial peer to the

From header in an incoming SIP Invite message.

URI voice class used to match the dial peer to the destination URI for an outgoing call.

Cisco IOS Voice Command Reference - S commands

441

show dialplan dialpeer

Field

modem transport huntstop in bound application associated out bound application associated dnis-map permission incoming COR list outgoing COR list

Translation profile (Incoming)

Translation profile (Outgoing) incoming call blocking translation-profile disconnect-cause voice-port type prefix forward-digits

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Cisco IOS Voice Command Reference - S commands show dial-peer through show gatekeeper zone prefix

Description

Transport method configured for modem calls. The default is system, which means that the value configured globally is used.

Whether the huntstop command is enabled or disabled for this dial peer.

IVR application that is associated with this dial peer when this dial peer is used for an inbound call leg.

IVR application that is associated with this dial peer when this dial peer is used for an outbound call leg.

Name of the dialed-number identification service

(DNIS) map that is configured in the dial peer with the dnis-map command.

Configured permission level for this dial peer.

Class of restriction (COR) criteria associated when matching an incoming dial peer.

COR criteria used to determine the appropriate outbound dial peer.

Incoming translation criteria applied to this dial peer.

Translation criteria applied to this dial peer when matching an outbound dial peer.

Indicates whether or not incoming call blocking has been applied for this dial peer.

The predefined translation profile associated with this dial peer.

Encoded network cause associated with the last call.

Voice port through which calls come into this dial peer.

Type of dial peer (POTS or VoIP).

Prefix number that is added to the front of the dial string before it is forwarded to the telephony device.

Which digits are forwarded to the telephony interface as configured using the forward-digits command.

show dial-peer through show gatekeeper zone prefix

Field

session-target direct-inward-dial digit_strip register E.164 number with GK fax rate payload size session-protocol req-qos acc-qos codec

Expect factor

Icpif

VAD voice class sip url

show dialplan dialpeer

Description

Configured session target (IP address or host name) for this dial peer.

Whether the direct-inward-dial command is enabled or disabled for this dial peer.

Whether digit stripping is enabled or disabled in the dial peer. Enabled is the default.

Indicates whether or not the dial peer has been configured to register its full E.164-format number with the local gatekeeper.

The transmission speed configured for fax calls. The default is system, which means that the value configured globally is used.

The size (in bytes) for a fax transmission payload.

Session protocol to be used for Internet calls between local and remote router via the IP backbone.

Configured requested quality of service for calls for this dial peer.

Lowest acceptable quality of service configured for calls for this dial peer.

Voice codec configured for this dial peer. Default is

G.729 (8 kbps).

User-requested expectation factor of voice quality for calls through this dial peer.

Configured calculated planning impairment factor

(ICPIF) value for calls sent by this dial peer.

Indicates whether or not voice activation detection

(VAD) is enabled for this dial peer.

URL format (SIP or TEL) used for SIP calls to this dial peer, as configured with the voice-class sip url command. The default is system, which means that the value configured globally with the url command in voice service VoIP SIP mode is used.

Cisco IOS Voice Command Reference - S commands

443

show dialplan dialpeer

Field

voice class sip rel1xx voice class perm tag

Connect Time

Charged Units

Successful Calls

Failed Calls

Incomplete Calls

Accepted Calls

Refused Calls

Last Disconnect Cause

Last Disconnect Text

Last Setup Time

Matched

Digits

Target

444

Cisco IOS Voice Command Reference - S commands show dial-peer through show gatekeeper zone prefix

Description

Indicates whether or not reliable provisional responses are supported, as configured with the voice-class sip

rel1xx command. The default is system, which means that the value configured globally with the rel1xx command in voice service VoIP SIP mode is used.

Voice class for a trunk that is assigned to this dial peer with the voice-class permanent command.

Unit of measure indicating the call connection time associated with this dial peer.

Number of call units charged to this dial peer.

Number of completed calls to this dial peer since system startup.

Number of uncompleted (failed) calls to this dial peer since system startup.

Number of incomplete calls to this dial peer since system startup.

Number of calls from this dial peer accepted since system startup.

Number of calls from this dial peer refused since system startup.

Encoded network cause associated with the last call.

This value is updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.

ASCII text describing the reason for the last call termination.

Value of the System Up Time when the last call to this peer was started.

Destination pattern matched for this dial peer.

Number of digits in this destination pattern matched for this dial peer.

Matched session target (IP address or host name) for this dial peer.

show dial-peer through show gatekeeper zone prefix show dialplan dialpeer

Related Commands

Command show dialplan in-carrier show dialplan in-trunk-group-label show dialplan incall show dialplan number

Description

Displays which VoIP or POTS dial peer is matched for a specific source carrier.

Displays which VoIP or POTS dial peer is matched for a specific source trunk group.

Displays which POTS dial peer is matched for a specific calling number or voice port.

Displays which dial peer is matched for a particular telephone number.

Cisco IOS Voice Command Reference - S commands

445

show dial-peer through show gatekeeper zone prefix show dialplan incall

show dialplan incall

To display which incoming POTS dial peer is matched for a specific calling number or voice port, use the

show dialplan incall number command in privileged EXEC mode.

show dialplan incall voice-port number calling-number [timeout]

Syntax Description

voice -port

calling -number

timeout

Voice port location. The syntax of this argument is platform-specific. For information on the syntax for a particular platform, see the voice-port command.

E.164 Calling number or ANI of the incoming voice call.

(Optional) Allows matching for variable-length destination patterns.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(1)T

12.2(8)T

Modification

This command was introduced on the Cisco 3600 series.

This command was implemented on the Cisco 1751, Cisco 2600 series, Cisco

3725, and Cisco 3745 and the timeout keyword was added.

Usage Guidelines

Use this command as a troubleshooting tool to determine which POTS dial peer is matched for an incoming call, for the selected calling number and voice port. The router attempts to match these items in the order listed:

1

Calling number with answer-address configured in dial peer

2

Calling number with destination-pattern configured in dial peer

3

Voice port with voice port configured in dial peer

The router first attempts to match a dial peer based on the calling number (ANI). If the router is unable to match a dial peer based on the calling number, it matches the call to a POTS dial peer based on the selected voice interface. If more than one dial peer uses the same voice port, the router selects the first matching dial peer. Use the timeout keyword to enable matching variable-length destination patters associated with dial peers. This can increase you r chances of finding a match for the dial peer number you specify.

446

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan incall

Examples

Note

For actual voice calls coming into the router, the router attempts to match the called number (the dialed number identification service [DNIS] number) with the incoming called-number configured in a dial peer.

The router, however, does not consider the called number when using the show dialplan incall number command.

The following sample output shows that an incoming call from interface 1/0/0:D with a calling number of

12345 is matched to POTS dial peer 10:

Router#

show dialplan incall 1/0/0:D number 12345

Macro Exp.: 12345

VoiceEncapPeer10 information type = voice, tag = 10, destination-pattern = `123..', answer-address = `', preference=0, numbering Type = `unknown' group = 10, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, huntstop = disabled, in bound application associated: DEFAULT out bound application associated: permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement type = pots, prefix = `', forward-digits default session-target = `', voice-port = `1/0/0:D', direct-inward-dial = disabled, digit_strip = enabled, register E.164 number with GK = TRUE

Connect Time = 0, Charged Units = 0, register E.164 number with GK = TRUE

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0,

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

Matched: 12345 Digits: 3

Target:

The following sample output shows that, if no dial peer has a destination pattern or answer address that matches the calling number of 888, the incoming call is matched to POTS dial peer 99, because the call comes in on voice port 1/0/1:D, which is the voice port configured for this dial peer:

Router#

show dialplan incall 1/0/1:D number 888

Macro Exp.: 888

VoiceEncapPeer99 information type = voice, tag = 99, destination-pattern = `99...', answer-address = `', preference=1, numbering Type = `national' group = 99, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, huntstop = disabled, in bound application associated: DEFAULT out bound application associated: permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement type = pots, prefix = `5', forward-digits 4

Cisco IOS Voice Command Reference - S commands

447

show dial-peer through show gatekeeper zone prefix show dialplan incall

session-target = `', voice-port = `1/0/1:D', direct-inward-dial = enabled, digit_strip = enabled, register E.164 number with GK = TRUE

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0,

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Matched:

Last Setup Time = 0.

Digits: 0

Target:

Related Commands

Command show dialplan dialpeer show dialplan in-carrier show dialplan in-trunk-group-label show dialplan number

Description

Displays which outbound dial peer is matched based upon the incoming dialed number and the COR criteria specified in the command line.

Displays which VoIP or POTS dial peer is matched for a specific source carrier.

Displays which VoIP or POTS dial peer is matched for a specific source trunk group.

Displays which dial peer is matched for a particular telephone number.

448

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan incall uri

show dialplan incall uri

To display which dial peer is matched for a specific uniform resource identifier (URI) in an incoming voice call, use the show dialplan incall uricommand in privileged EXEC mode.

H.323 Session Protocol

show dialplan incall uri h323 {called| calling} uri

SIP Session Protocol

show dialplan incall uri sip {from| request| to} uri

Syntax Description called calling from request to

uri

Voice class that is configured in dial peers with the

incoming uri called command.

Voice class that is configured in dial peers with the

incoming uri callingcommand.

Voice class that is configured in dial peers with the

incoming uri fromcommand.

Voice class that is configured in dial peers with the

incoming uri requestcommand.

Voice class that is configured in dial peers with the

incoming uri tocommand.

URI of the incoming call.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(4)T

Modification

This command was introduced.

Usage Guidelines

• Use this command for troubleshooting to determine which dial peer is matched for an incoming call, based on the selected URI and the specified field in the call message.

Cisco IOS Voice Command Reference - S commands

449

show dial-peer through show gatekeeper zone prefix show dialplan incall uri

Examples

• To set the URI format for matching calls, use the voice class uri command. To set the URI voice class in the inbound dial peer, use the incoming uri command.

The following is sample output from this command for a SIP URI:

Router#

show dialplan incall uri sip from sip:5551234

Inbound VoIP dialpeer matching based on SIP URI's

VoiceOverIpPeer10 peer type = voice, information type = voice, description = `', tag = 10, destination-pattern = `', answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 10, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, modem transport = system,

URI classes:

Incoming (Request) =

Incoming (To) =

Incoming (From) = 101

Destination = huntstop = disabled, in bound application associated: 'get_headers_tcl' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' type = voip, session-target = `', technology prefix: settle-call = disabled ip media DSCP = ef, ip signaling DSCP = af31, UDP checksum = disabled, session-protocol = sipv2, session-transport = system, req-qos = best-ef acc-qos = best-effort,

RTP dynamic payload type values: NTE = 101

Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122

CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8

RTP comfort noise payload type = 19 fax rate = voice, payload size = 20 bytes fax protocol = system fax-relay ecm enable fax NSF = 0xAD0051 (default) codec = g729r8, payload size = 20 bytes,

Expect factor = 0, Icpif = 20,

Playout Mode is set to default,

Initial 60 ms, Max 300 ms

Playout-delay Minimum mode is set to default, value 40 ms

Fax nominal 300 ms

Max Redirects = 1, signaling-type = ext-signal,

VAD = enabled, Poor QOV Trap = disabled,

Source Interface = NONE voice class sip url = system, voice class sip rel1xx = system, voice class perm tag = `'

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

450

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan incall uri

Last Disconnect Cause is "",

Last Disconnect Text is "",

Matched:

Last Setup Time = 0.

Digits: 0

Target:

The following is sample output from this command for a TEL URI:

Router#

show dialplan incall uri h323 called tel:1234567

Inbound VoIP dialpeer matching based on H323 URI's

VoiceOverIpPeer25 peer type = voice, information type = voice, description = `', tag = 25, destination-pattern = `', answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 25, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, modem transport = system,

URI classes:

Incoming (Called) = 103

Incoming (Calling) =

Destination = huntstop = disabled, in bound application associated: 'callme' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' type = voip, session-target = `ipv4:10.10.1.1', technology prefix: settle-call = disabled ip media DSCP = ef, ip signaling DSCP = af31, UDP checksum = disabled, session-protocol = cisco, session-transport = system, req-qos = best-ef acc-qos = best-effort,

RTP dynamic payload type values: NTE = 101

Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122

CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8

RTP comfort noise payload type = 19 fax rate = voice, payload size = 20 bytes fax protocol = system fax-relay ecm enable fax NSF = 0xAD0051 (default) codec = g729r8, payload size = 20 bytes,

Expect factor = 0, Icpif = 20,

Playout Mode is set to default,

Initial 60 ms, Max 300 ms

Playout-delay Minimum mode is set to default, value 40 ms

Fax nominal 300 ms

Max Redirects = 1, signaling-type = ext-signal,

VAD = enabled, Poor QOV Trap = disabled,

Source Interface = NONE voice class sip url = system, voice class sip rel1xx = system, voice class perm tag = `'

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Cisco IOS Voice Command Reference - S commands

451

show dial-peer through show gatekeeper zone prefix show dialplan incall uri

Matched:

Last Setup Time = 0.

Digits: 0

Target:

The table below describes significant fields in the displays.

Table 68: show dialplan incall uri Field Descriptions

Field

VoiceOverIpPeer information type tag destination-pattern answer-address preference

Admin state

Operation state incoming called-number

DTMF Relay

URI classes: Incoming (Request)

URI classes: Incoming (To)

URI classes: Incoming (From)

URI classes: Destination huntstop in bound application associated

Description

Dial peer associated with the calling number entered.

Information type for this call; for example, voice or fax.

Unique number that identifies the dial peer.

Destination pattern (called number) configured for this dial peer.

Answer address (calling number) configured for this dial peer.

Hunt group preference order set for this dial peer.

Administrative state of this dial peer.

Operational state of this dial peer.

Called number (DNIS) configured for this dial peer.

Whether the dtmf-relay command is enabled or disabled for this dial peer.

URI voice class used for matching dial peer to

Request-URI in an incoming SIP Invite message.

URI voice class used for matching dial peer to the To header in an incoming SIP Invite message.

URI voice class used for matching dial peer to the

From header in an incoming SIP Invite message.

URI voice class used to match the dial peer to the destination URI for an outgoing call.

Whether the huntstop command is enabled or disabled for this dial peer.

IVR application that is associated with this dial peer when this dial peer is used for an inbound call leg.

452

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Field

out bound application associated dnis-map permission type session-target session-protocol req-qos acc-qos codec

Expect factor

Icpif

VAD voice class sip url voice class sip rel1xx voice class perm tag

show dialplan incall uri

Description

IVR application that is associated with this dial peer when this dial peer is used for an outbound call leg.

Name of the dialed-number identification service

(DNIS) map that is configured in the dial peer with the dnis-map command.

Configured permission level for this peer.

Type of dial peer (POTS or VoIP).

Configured session target (IP address or host name) for this dial peer.

Session protocol to be used for Internet calls between local and remote router via the IP backbone.

Configured requested quality of service for calls for this dial peer.

Lowest acceptable quality of service configured for calls for this peer.

Voice codec configured for this dial peer. Default is

G.729 (8 kbps).

User-requested expectation factor of voice quality for calls through this peer.

Configured calculated planning impairment factor

(ICPIF) value for calls sent by a dial peer.

Whether voice activation detection (VAD) is enabled for this dial peer.

URL format (SIP or TEL) used for SIP calls to this dial peer, as configured with the voice-class sip url command. The default is system, which means that the value configured globally with the url command in voice service VoIP SIP mode is used.

Whether reliable provisional responses are supported, as configured with the voice-class sip rel1xx command. The default is system, which means that the value configured globally with the rel1xx command in voice service VoIP SIP mode is used.

Voice class for a trunk that is assigned to this dial peer with the voice-class permanent command.

Cisco IOS Voice Command Reference - S commands

453

show dialplan incall uri

Field

Connect Time

Charged Units

Successful Calls

Failed Calls

Accepted Calls

Refused Calls

Last Disconnect Cause

Last Disconnect Text

Last Setup Time

Matched

Target

Related Commands

Command debug voice uri incoming uri session protocol show dial-peer voice

454

Cisco IOS Voice Command Reference - S commands show dial-peer through show gatekeeper zone prefix

Description

Unit of measure indicating the call connection time associated with this dial peer.

Number of call units charged to this dial peer.

Number of completed calls to this peer since system startup.

Number of uncompleted (failed) calls to this peer since system startup.

Number of calls from this peer accepted since system startup.

Number of calls from this peer refused since system startup.

Encoded network cause associated with the last call.

This value is updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.

ASCII text describing the reason for the last call termination.

Value of the System Up Time when the last call to this peer was started.

Destination pattern matched for this dial peer.

Matched session target (IP address or host name) for this dial peer.

Description

Displays debugging messages related to URI voice classes.

Specifies the voice class used to match a VoIP dial peer to the URI of an incoming call.

Specifies the session protocol in the dial peer for calls between the local and remote router.

Displays detailed and summary information about voice dial peers.

show dial-peer through show gatekeeper zone prefix

Command show dialplan uri voice class uri voice class uri sip preference show dialplan incall uri

Description

Displays which outbound dial peer is matched for a specific destination URI.

Creates or modifies a voice class for matching dial peers to calls containing a SIP or TEL URI.

Sets a preference for selecting voice classes for a SIP

URI.

Cisco IOS Voice Command Reference - S commands

455

show dial-peer through show gatekeeper zone prefix show dialplan in-carrier

show dialplan in-carrier

To display which incoming VoIP or POTS dial peer is matched for a specific source carrier or voice port, use the show dialplan in-carrier command in privileged EXEC mode.

show dialplan in-carrier carrier-id [voip| pots]

Syntax Description

carrier -id

voip pots

VoIP or POTS source carrier identifier.

(Optional) Allows you to limit the search criteria to only VoIP dial peers.

(Optional) Allows you to limit the search criteria to only POTS dial peers.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(13)T

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on Cisco AS5300, Cisco AS5400, and Cisco AS5800 access servers.

Usage Guidelines

Use this command as a troubleshooting tool to determine which VoIP or POTS dial peer is matched for an incoming call, based on the carrier identifier specified in the command line. Use the voip or pots keywords to further limit the scope of possible matches for the dial peer specified in the show dialplan command line.

Examples

The following sample output shows a VoIP or POTS dial peer being matched to another POTS dial peer based on its carrier identifier, "aaa":

Router# show dialplan in-carrier aaa pots

Inbound pots dialpeer Matching based on source carrier-id

VoiceEncapPeer7777 information type = voice, description = `', tag = 7777, destination-pattern = `', answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent source carrier-id = `aaa', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 7777, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, huntstop = disabled,

456

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan in-carrier

in bound application associated:'DEFAULT' out bound application associated:'' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' voice-port = `' type = pots, prefix = `', forward-digits default session-target = `', up, direct-inward-dial = disabled, digit_strip = enabled, register E.164 number with GK = TRUE fax rate = system, payload size = 20 bytes

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Setup Time = 0.

Matched: Digits:0

Target:

Last Disconnect Cause is "",

Last Disconnect Text is "",

Related Commands

Command show dialplan dialpeer show dialplan incall show dialplan in-trunk-group-label show dialplan number

Description

Displays which outbound dial peer is matched based upon the incoming dialed number and the COR criteria specified in the command line.

Displays which POTS dial peer is matched for a specific calling number or voice port.

Displays which VoIP or POTS dial peer is matched for a specific source trunk group.

Displays which dial peer is matched for a particular telephone number.

Cisco IOS Voice Command Reference - S commands

457

show dial-peer through show gatekeeper zone prefix show dialplan in-trunk-group-label

show dialplan in-trunk-group-label

To display which incoming VoIP or POTS dial peer is matched for a specific trunk group label, use the show

dialplan in-trunk-group-label command in privileged EXEC mode.

show dialplan in-trunk-group-label trunk-group-label {pots| voip}

Syntax Description

trunk -group-label

voip pots

VoIP or POTS source trunk group identifier.

(Optional) Allows you to limit the search criteria to only VoIP dial peers.

(Optional) Allows you to limit the search criteria to only POTS dial peers.

Command Modes

Privileged EXEC (#)

Command History

Release

12.2(13)T

Modification

This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on Cisco AS5300, Cisco AS5400, and Cisco AS5800 access servers.

Usage Guidelines

Use this command to determine which VoIP or POTS dial peer is matched for an incoming call, based on the identifier of the source trunk group. The router attempts to match these items in the order listed. Use the voip or pots keywords to further limit the scope of possible matches for the dial peer specified in the show dialplan command line.

Examples

The following sample output shows an inbound VoIP or POTS dial peer being matched to an outbound POTS dial peer based on the trunk group label "NYtrunk":

Router# show dialplan in-trunk-group-label NYtrunk pots

Inbound pots dialpeer Matching based on source trunk-group-label

VoiceEncapPeer2003 information type = voice, description = `', tag = 2003, destination-pattern = `', answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent source carrier-id = `', target carrier-id = `', source trunk-group-label = `NYtrunk', target trunk-group-label = `', numbering Type = `unknown' group = 2003, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled,

458

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan in-trunk-group-label

huntstop = disabled, in bound application associated:'debit-card' out bound application associated:'' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' voice-port = `' type = pots, prefix = `', forward-digits default session-target = `', up, direct-inward-dial = disabled, digit_strip = enabled, register E.164 number with GK = TRUE fax rate = system, payload size = 20 bytes

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

Matched: Digits:0

Target:

Related Commands

Command show dialplan dialpeer show dialplan in-carrier show dialplan incall show dialplan number

Description

Displays which outbound dial peer is matched based upon the incoming dialed number and the COR criteria specified in the command line.

Displays which VoIP or POTS dial peer is matched for a specific source carrier.

Displays which POTS dial peer is matched for a specific calling number or voice port.

Displays which dial peer is matched for a particular telephone number.

Cisco IOS Voice Command Reference - S commands

459

show dial-peer through show gatekeeper zone prefix show dialplan number

show dialplan number

To display which outgoing dial peer is reached when a particular telephone number is dialed, use the show

dialplan number command in privileged EXEC mode.

show dialplan number dial-string [carrier identifier] [fax| huntstop| voice] [timeout]

Syntax Description

dial -string

carrier

identifier fax

huntstop

timeout voice

Particular destination pattern (E.164 telephone number).

(Optional) Indicates that you wish to base your search for applicable dial peers on the source carrier identifier.

(Optional) Source carrier identifier to accompany the

carrier keyword.

(Optional) Fax information type.

(Optional) Terminates further dial-peer hunting upon encountering the first dial-string match.

(Optional) Allows matching for variable-length destination patterns.

(Optional) Voice information type.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(1)T

12.2(1)

12.2(8)T

12.2(11)T

Modification

This command was introduced on the Cisco 3600 series.

The huntstop keyword was added.

This command was implemented on the Cisco 1751, Cisco 2600 series,

Cisco 3725, and Cisco 3745 and the timeout keyword was added.

The carrier, fax, and voice keywords were added.

460

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan number

Usage Guidelines

Use this command to test whether the dial plan configuration is valid and working as expected. Use the timeout keyword to enable matching variable-length destination patters associated with dial peers. This can increase you r chances of finding a match for the dial peer number you specify.

Examples

The following is sample output from thiscommand using a destination pattern of 1001:

Router#

show dialplan number 1001

Macro Exp.: 1001

VoiceEncapPeer1003 information type = voice, tag = 1003, destination-pattern = `1001', answer-address = `', preference=0, numbering Type = `unknown' group = 1003, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, huntstop = enabled, type = pots, prefix = `', forward-digits default session-target = `', voice-port = `1/1', direct-inward-dial = disabled,

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

Matched: 1001 Digits: 4

Target:

VoiceEncapPeer1004 information type = voice, tag = 1004, destination-pattern = `1001', answer-address = `', preference=0, numbering Type = `unknown' group = 1004, Admin state is up, Operation state is up,

...

Matched: 1001 Digits: 4

Target:

VoiceEncapPeer1002 information type = voice, tag = 1002, destination-pattern = `1001', answer-address = `', preference=0, numbering Type = `unknown' group = 1002, Admin state is up, Operation state is up,

...

Matched: 1001 Digits: 4

Target:

VoiceEncapPeer1001 information type = voice, tag = 1001, destination-pattern = `1001', answer-address = `', preference=0,

...

Matched: 1001 Digits: 4

Target: numbering Type = `unknown' group = 1001, Admin state is up, Operation state is up,

The following is sample output from this command using a destination pattern of 1001 and the huntstop keyword:

Router#

show dialplan number 1001 huntstop

Macro Exp.: 1001

VoiceEncapPeer1003 information type = voice, tag = 1003, destination-pattern = `1001', answer-address = `', preference=0,

Cisco IOS Voice Command Reference - S commands

461

show dial-peer through show gatekeeper zone prefix show dialplan number

numbering Type = `unknown' group = 1003, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, huntstop = enabled, type = pots, prefix = `', forward-digits default session-target = `', voice-port = `1/1', direct-inward-dial = disabled,

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Setup Time = 0.

Matched: 1001 Digits: 4

Target:

Last Disconnect Cause is "",

Last Disconnect Text is "",

Related Commands

Command show dialplan dialpeer show dialplan incall show dialplan in-carrier show dialplan in-trunk-group-label

Description

Displays which outbound dial peer is matched based upon the incoming dialed number and the COR criteria specified in the command line.

Displays which POTS dial peer is matched for a specific calling number or voice port.

Displays which VoIP or POTS dial peer is matched for a specific source carrier.

Displays which VoIP or POTS dial peer is matched for a specific source trunk group.

462

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dialplan uri

show dialplan uri

To display which outbound dial peer is matched for a specific destination uniform resource identifier (URI), use the show dialplan uricommand in privileged EXEC mode.

show dialplan uri uri

Syntax Description

uri

Destination Session Initiation Protocol (SIP) or telephone (TEL) URI for the outgoing call.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(4)T

Modification

This command was introduced.

Usage Guidelines

Examples

• Use this command for troubleshooting to determine which dial peer is matched for an outgoing call, based on the selected URI.

• To set the URI format used to match calls, use the voice class uri command. To set the URI voice class in the outbound dial peer, use the destination uri command.

The following is sample output from this command:

Router#

show dialplan uri sip:123456

Outbound dialpeer matching based on destination URI

VoiceOverIpPeer99 peer type = voice, information type = voice, description = `', tag = 99, destination-pattern = `', answer-address = `', preference=0,

CLID Restriction = None

CLID Network Number = `'

CLID Second Number sent source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 99, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled, modem transport = system,

URI classes:

Incoming (Request) =

Cisco IOS Voice Command Reference - S commands

463

show dial-peer through show gatekeeper zone prefix show dialplan uri

Incoming (To) =

Incoming (From) =

Destination = 100 huntstop = disabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

Translation profile (Incoming):

Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' type = voip, session-target = `', technology prefix: settle-call = disabled ip media DSCP = ef, ip signaling DSCP = af31, UDP checksum = disabled, session-protocol = sipv2, session-transport = system, req-qos = best-ef acc-qos = best-effort,

RTP dynamic payload type values: NTE = 101

Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122

CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8

RTP comfort noise payload type = 19 fax rate = voice, payload size = 20 bytes fax protocol = system fax-relay ecm enable fax NSF = 0xAD0051 (default) codec = g729r8, payload size = 20 bytes,

Expect factor = 0, Icpif = 20,

Playout Mode is set to default,

Initial 60 ms, Max 300 ms

Playout-delay Minimum mode is set to default, value 40 ms

Fax nominal 300 ms

Max Redirects = 1, signaling-type = ext-signal,

VAD = enabled, Poor QOV Trap = disabled,

Source Interface = NONE voice class sip url = system, voice class sip rel1xx = system, voice class perm tag = `'

Time elapsed since last clearing of voice call statistics never

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Matched:

Target:

Last Setup Time = 0.

Digits: 0

Related Commands

Command debug voice uri destination uri show dialplan incall uri voice class uri

Description

Displays debugging messages related to URI voice classes.

Specifies the voice class used to match the dial peer to the destination URI for an outgoing call.

Displays which dial peer is matched for a specific

URI in an incoming call.

Creates or modifies a voice class for matching dial peers to a SIP or TEL URI.

464

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Command voice class uri sip preference show dialplan uri

Description

Sets a preference for selecting voice classes for a SIP

URI.

Cisco IOS Voice Command Reference - S commands

465

show dial-peer through show gatekeeper zone prefix show dn-numbers

show dn-numbers

To display directory number information of Call Manager Express (CME), use the show dn-numbers command in user EXEC or privileged EXEC mode.

show dn-numbers

Syntax Description

This command has no arguments or keywords.

Command Modes

User EXEC (>) Privileged EXEC (#))

Command History

Release

12.4(15)T

Cisco IOS XE Release 2.4

Modification

This command was introduced.

This command was integrated into Cisco IOS XE Release 2.4.

Examples

The following is sample output from the show dn-numbers command:

Router#

show dn-numbers

Directory numbers

Entry

1 name user1

10

3 user2 user3

4

12

11 user4 user5 user6 number

0

7890

1234

890

5676

987 ephone directory numbers

DN name

2

4

6

10 user7 user10 user11 user12 sip phone numbers

DN name

1

8

9 user13 user14 user15 number

1000

34567

1234567891

1234567 number

10000

87953893

Not Configured

The table below describes the significant fields shown in the display.

Table 69: show dn-numbers Field Descriptions

Field

DN name

Description

Directory number.

Name of the connection.

466

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Field

number

Description

Telephone number.

show dn-numbers

Cisco IOS Voice Command Reference - S commands

467

show dial-peer through show gatekeeper zone prefix show dspfarm

show dspfarm

To display digital signal processor (DSP) farm service information such as operational status and DSP resource allocation for transcoding and conferencing, use the show dspfarm command in user EXEC or privileged

EXEC mode.

show dspfarm [all| dsp {active| all| idle| stats bridge-id [sample seconds]}| profile [ profile-id ]| sessions

[ session-id ]| video {conference| statistics| transcode}]

Cisco ASR 1000 Series Router

show dspfarm {all| dsp {active| all| idle| stats bridge-id [sample seconds]}| profile [ profile-identifier ]}

Syntax Description all dsp active all idle stats

bridge-id

sample

seconds

profile

profile-id

sessions

session-id

(Optional) Displays all global information about the

DSP farm service.

(Optional) Displays DSP information about the DSP farm service.

Displays active DSP information about the DSP farm service.

Displays all DSP information about the DSP farm service.

Displays idle DSP information about the DSP farm service.

Displays DSP statistics about the DSP farm service.

Displays the DSP statistics for a call bridge the specified bridge ID.

(Optional) Displays statistics of the specified sample interval.

(Optional) The DSP sample interval time, in seconds.

(Optional) Displays profiles about the DSP farm service.

(Optional) The profile ID about the DSP farm service.

(Optional) Displays sessions and connections about the DSP farm service.

(Optional) The session identifier to be displayed for the DSP farm service.

468

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dspfarm video conference statistics transcode

(Optional) Displays information on video resources.

(Optional) Displays the DSP information, such as the codecs, video bridge channel, and transmit (tx) and receive (rx) packets that are used for each participant in a conference and is grouped by conference sessions.

(Optional) Displays the DSP statistics of the call bridge.

(Optional) Displays the DSP status and statistics for the transcoding call.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

12.1(5)YH

12.2(13)T

12.4(15)T

Cisco IOS XE Release 3.2S

15.1(4)M

Modification

This command was introduced on the Cisco VG200.

This command was implemented on the Cisco 2600 series, Cisco 3620,

Cisco 3640, Cisco 3660, and Cisco 3700 series.

The stats, sample, sessions,and profile keywords were added. The bridge-id,

profile-id, seconds, and session-id arguments were added.

This command was implemented on the Cisco ASR 1000 Series Router.

This command was modified. The video, conference, statistics, and

transcode keywords were added.

Usage Guidelines

The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms

(NM-HDV-FARMs) to provide DSP resources.

Cisco ASR 1000 Series Router

The show dspfarm command is used to view the DSP farm service information such as operational status and

DSP resource allocation for transcoding.

Note

The session keyword and session-id argument is not supported on Cisco ASR 1000 Series Router.

Cisco IOS Voice Command Reference - S commands

469

show dial-peer through show gatekeeper zone prefix show dspfarm

Examples

The following is sample output from several forms of the show dspfarmcommand. The fields are self explanatory.

Router#

show dspfarm

DSPFARM Configuration Information:

Admin State: UP, Oper Status: ACTIVE - Cause code: NONE

Transcoding Sessions: 4, Conferencing Sessions: 0

RTP Timeout: 600

Router#

show dspfarm all

DSPFARM Configuration Information:

Admin State: UP, Oper Status: ACTIVE - Cause code: NONE

Transcoding Sessions: 4, Conferencing Sessions: 2

RTP Timeout: 1200

Connection average duration: 3600, Connection check interval 600

Codec G729 VAD: ENABLED

1

1

1

1

1

1

1

1

1

1

1

1

1

1

1

1

1

1

Total number of active session(s) 0, and connection(s) 0

SLOT DSP CHNL STATUS USE TYPE SESS-ID CONN-ID PKTS-RXED PKTS-TXED

1 3 1 UP FREE conf -

3

3

3

3

3

4

4

2

3

4

5

6

1

2

UP

UP

UP

UP

UP

UP

UP

FREE

FREE

FREE

FREE

FREE

FREE

FREE conf conf conf conf conf conf conf

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

4

4

4

4

5

5

5

5

5

5

5

3

4

5

6

1

2

3

4

5

6

7

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

FREE

FREE

FREE

FREE

FREE

FREE

FREE

FREE conf conf conf conf xcode xcode xcode xcode

-

-

-

-

-

-

-

-

FREE xcode -

FREE xcode -

FREE xcode -

1 5 8 UP FREE xcode -

Total number of DSPFARM DSP channel(s) 20

Router#

show dspfarm dsp all

DSPFARM Configuration Information:

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

1

1

1

1

1

1

1

1

1

1

1

1

1

Admin State: UP, Oper Status: ACTIVE - Cause code: NONE

Transcoding Sessions: 4, Conferencing Sessions: 2

RTP Timeout: 1200

Connection average duration: 3600, Connection check interval 600

Codec G729 VAD: ENABLED

1

1

1

Total number of active session(s) 0, and connection(s) 0

SLOT DSP CHNL STATUS USE TYPE SESS-ID CONN-ID PKTS-RXED PKTS-TXED

1

1

3

3

1

2

UP

UP

FREE

FREE conf conf

-

-

-

-

-

-

-

-

3

3

3

3

4

5

UP

UP

UP

FREE conf -

FREE conf -

FREE conf -

-

-

-

-

-

-

-

-

-

1

1

3

4

4

4

4

4

4

5

5

5

5

5

5

5

5

6

1

2

3

4

5

6

1

2

3

4

5

6

7

8

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

UP

FREE conf -

FREE conf -

FREE conf -

FREE conf -

FREE conf -

FREE conf -

FREE

FREE xcode -

FREE xcode -

FREE xcode -

FREE xcode -

FREE xcode -

FREE xcode -

FREE

FREE conf xcode xcode

-

-

-

Total number of DSPFARM DSP channel(s) 20

Router#

show dspfarm sessions

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

470

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dspfarm

sess_id conn_id stype mode

4 145 xcode sendrecv codec g711a pkt

20 ripaddr

10.10.10.19

rport

19460 sport

21284

4

5

5

161

177

193 xcode xcode xcode sendrecv sendrecv sendrecv g729 g711u g729b

10

20

10

10.10.10.28

10.10.10.17

10.10.10.18

19414

18290

19150

20382

21170

18968

The following sample output displays dspfarm profiles for video conferencing and video transcoding.

Router#

show dspfarm profile

Profile ID = 1, Service = VIDEO CONFERENCING, Resource ID = 2

Video Conference Type : HOMOGENEOUS, Layout : disabled

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : DOWN

Profile Operation State : DOWN

Application : SCCP Status : NOT ASSOCIATED

Resource Provider : FLEX_DSPRM Status : NONE

Number of Resource Configured : 1

Number of Resource Available : 0

Maximum conference participants : 16

Codec Configuration: num_of_codecs:6

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

Codec : g729r8, Maximum Packetization Period : 60

Codec : g729br8, Maximum Packetization Period : 60

Video Codec Configuration:

Codec : h263

Resolution : cif

Frame rate:30, Min bitrate:320kbps, Max bitrate:320kbps

Payload protocol : rfc-2190, Extension : annex-none

Profile ID = 2, Service = VIDEO CONFERENCING, Resource ID = 3

Video Conference Type : HETEROGENEOUS, Layout : disabled

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE IN PROGRESS

Application : SCCP Status : ASSOCIATION IN PROGRESS

Resource Provider : FLEX_DSPRM Status : UP

Number of Resource Configured : 1

Number of Resource Available : 1

Maximum conference participants : 4

Maximum video ports : 4

Codec Configuration: num_of_codecs:6

Codec : g729br8, Maximum Packetization Period : 60

Codec : g729r8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g711ulaw, Maximum Packetization Period : 30

Video Codec Configuration:

Codec : h264

Resolution : qcif

Frame rate:15, Min bitrate:64kbps, Max bitrate:704kbps

Frame rate:30, Min bitrate:64kbps, Max bitrate:704kbps

Resolution : cif

Frame rate:15, Min bitrate:64kbps, Max bitrate:704kbps

Frame rate:30, Min bitrate:64kbps, Max bitrate:704kbps

Codec : h263

Resolution : qcif

Frame rate:15, Min bitrate:64kbps, Max bitrate:704kbps

Frame rate:30, Min bitrate:64kbps, Max bitrate:704kbps

Resolution : cif

Frame rate:15, Min bitrate:64kbps, Max bitrate:704kbps

Frame rate:30, Min bitrate:64kbps, Max bitrate:704kbps

Dspfarm Profile Configuration

Profile ID = 3, Service =Universal TRANSCODING, Resource ID = 1

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : DOWN

Profile Operation State : DOWN

Cisco IOS Voice Command Reference - S commands

471

show dial-peer through show gatekeeper zone prefix show dspfarm

Application : SCCP Status : NOT ASSOCIATED

Resource Provider : FLEX_DSPRM Status : NONE

Number of Resource Configured : 0

Number of Resource Available : 0

Codec Configuration: num_of_codecs:4

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

The following sample output displays DSP information for video conferences.

Router#

show dspfarm video conference

VIDEO CONFERENCE SESSION: slot 0 dsp 3 channel_id 1 rsc_id 8 profile_id 101 conferee_id 1 name_num: 62783363 audio_codec g711u pkt_size 160 bridge_id 1 dsp_txed_pkts 25993 dsp_rxed_pkts 25888 conferee_id 1 name_num: 62783363 video_codec H264_VGA rfc_number RFC3984 payload rx: 97 tx:97 framerate 30 bitrate(k) 960 annex 0x40 cluster_id 0 bridge_id 2 dsp_txed_pkts 59230 layout_id 0 dsp_rxed_pkts 63019 conferee_id 2 name_num: 62783365 audio_codec g711u dsp_txed_pkts 21682 pkt_size 160 bridge_id 3 dsp_rxed_pkts 21598 conferee_id 2 name_num: 62783365 video_codec H264_4CIF rfc_number RFC3984 payload rx: 97 tx:97 framerate 30 bitrate(k) 960 annex 0x40 cluster_id 1 bridge_id 4 layout_id 0 dsp_txed_pkts 49488 dsp_rxed_pkts 78510 conferee_id 3 name_num: 3004 audio_codec g711u pkt_size 160 bridge_id 5 dsp_txed_pkts 12130 conferee_id 3 name_num: 3004 dsp_rxed_pkts 12067 video_codec H264_CIF rfc_number RFC3984 payload rx: 97 tx:97 framerate 30 bitrate(k) 704 annex 0x40 cluster_id 2 bridge_id 6 dsp_txed_pkts 20354 layout_id 0 dsp_rxed_pkts 25702 conferee_id 4 name_num: LifeSize LifeSize audio_codec g711u pkt_size 160 bridge_id 7 dsp_txed_pkts 1751 dsp_rxed_pkts 1672 conferee_id 4 name_num: LifeSize LifeSize video_codec H264_4CIF rfc_number RFC3984 payload rx: 96 tx:96 framerate 30 bitrate(k) 1100 annex 0x40 cluster_id 1 bridge_id 8 layout_id 0 dsp_txed_pkts 3558 dsp_rxed_pkts 3569 cluster_id 0 video_codec H264_VGA rfc_number RFC3984 rfc_payload 100 framerate 30 bitrate(k) 1000, annex 0x40 decoder_id 1 slot 0 dsp 13 codec h264 vga encoder_id 1 slot 0 dsp 10 codec h264 vga cluster_id 0 cluster_id 0 cluster_id 1 video_codec H264_4CIF rfc_number RFC3984 rfc_payload 100 framerate 30 bitrate(k) 1000, annex 0x40 decoder_id 1 slot 0 dsp 2 codec h264 4cif encoder_id 1 slot 0 dsp 7 codec h264 4cif cluster_id 1 cluster_id 1 cluster_id 2 video_codec H264_CIF rfc_number RFC3984 rfc_payload 100 framerate 30 bitrate(k) 704 , annex 0x40 decoder_id 1 slot 0 dsp 15 codec h264 cif encoder_id 1 slot 0 dsp 14 codec h264 cif

Total number of DSPFARM DSP channel(s) 1 cluster_id 2 cluster_id 2

The following sample output displays the statistics for a call that uses video transcoding.

Router#

show dspfarm dsp stats

Gathering total stats...

Video Statistics for bridge_id=3 call_id=2

Video Decoder Statistics:

Slot=0 DSP_Id=8 Decoder_Id=1

CallDuration=268 Codec=1 ProfileId=0x0 LevelId=0

PicWidth=352 PicHeight=288 FrameRate=30 Bitrate=360000

NumMacroBlocksConcealed=0 NumFramesConcealed=0

NumPackets=13269 NumBytesConsumed=12096254

NumBadHeaderPackets=0 NumOutOfSyncPackets=24

472

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dspfarm

NumBufferOverflow=0

Video Encoder Statistics:

Slot=0 DSP_Id=2 Encoder_Id=1

Duration=268 Codec=1 ProfileId=0x0 LevelId=0

PicWidth=176 PicHeight=144 FrameRate=30 Bitrate=704000

InstantBitrate=440000 NumPackets=17571 NumBytesGenerated=14830996

The following sample output displays the statistics for a video conference.

Router#

show dspfarm dsp stats

Gathering total stats...

Video Statistics for bridge_id=3 call_id=4

Video Conferee Status - ConfereeID=1

ContributionState=0x1 IngressMute=0 EgressMute=0

DtmfRtpPlt=0 ClusterId=1 StreamDir=3

PayloadType=0x6161 TxSSRC=0x1F3C RtpProtocol=2

CodecType=2 Annex=0x0 PicWidth=352 PicHeight=288

FrameRate=30 Bitrate(x100)=3760

Video Conferee Statistics - ConfereeID=1

TotalRxPackets=5076 TotalRxBytes=3957126

TotalTxPackets=3829 TotalTxBytes=3429797

TotalDroppedPackets=3 CurDroppedPackets=0

TotalOutOfOrderPackets=0 CurOutOfOrderPackets=0

MaxObservedJitter=0 CurObservedJitter=0

MaxObservedDelay=0 CurObservedDelay=0

MaxOutOfSyncDelay=0 CurOutOfSyncDelay=0

ActualFrameRate=0 ActualBitrate(x100)=2017

FastVideoUpdateRate=0 TotalDuration=135

Video Conference Status:

ServiceType=0 MuteAllStatus=0

CurSpeakerConfereeId=1 LastSpeakerConfereeId=3 NewSpeakerConfereeId=0

ConfereeIdBitMap=0x07

Video Conference Statistics:

NumActiveChans=3 NumMaxChans=1

TotalRxPackets=42589 TotalRxBytes=29979147

TotalTxPackets=12361 TotalTxBytes=10003701

TotalDroppedPackets=3 CurDroppedPackets=0

TotalOutOfOrderPackets=0 CurOutOfOrderPackets=0

MaxObservedJitter=0 CurObservedJitter=0

MaxObservedDelay=0 CurObservedDelay=0

MaxOutOfSyncDelay=0 CurOutOfSyncDelay=0

The following is sample output of the show dspfarmall command on Cisco ASR 1000 Series Router.

Router#

show dspfarm all

Dspfarm Profile Configuration

Profile ID = 1, Service = TRANSCODING, Resource ID = 1

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SBC Status : ASSOCIATED

Resource Provider : FLEX_DSPRM Status : UP

Number of Resources Configured : 588

Number of Resources Out of Service : 0

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

SLOT DSP VERSION STATUS CHNL USE TYPE

5 1 26.7.0

UP N/A FREE xcode 1

RSC_ID BRIDGE_ID

-

5

5

5

5

1

1

1

1

26.7.0

26.7.0

26.7.0

26.7.0

UP

UP

UP

UP

N/A

N/A

N/A

N/A

FREE

FREE

FREE

FREE xcode xcode xcode xcode

1

1

1

1

-

-

-

-

-

-

-

-

-

-

-

-

-

The following is sample output of the

show dspfarm

dsp idle command providing idle dsp information on Cisco ASR 1000 Series Router.

Router#

show dspfarm dsp idle

SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID

Cisco IOS Voice Command Reference - S commands

473

show dial-peer through show gatekeeper zone prefix show dspfarm

5

5

5

5

5

5

5

5

5

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

1 26.7.0

UP

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

N/A FREE xcode 1

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

-

5 1 26.7.0

UP N/A FREE xcode 1

The following is sample output of the

show dspfarm

profile 1 command providing DSP Farm profile configuration details such as application association, number of resources configured, Codecs added, and maximum number of sessions for profile 1 on Cisco ASR 1000 Series Router.

Router#

show dspfarm profile 1

Dspfarm Profile Configuration

Profile ID = 1, Service = TRANSCODING, Resource ID = 1

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SBC Status : ASSOCIATED

Resource Provider : FLEX_DSPRM Status : UP

Number of Resources Configured : 588

Number of Resources Out of Service : 0

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

Router#show dspfarm profile ?

<1-65535> Profile ID

|

<cr>

Output modifiers

Related Commands

Command dspfarm (DSP farm)

Description

Enables DSP-farm service.

474

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dspfarm profile

show dspfarm profile

To display configured digital signal processor (DSP) farm profile information for a selected Cisco CallManager group, use the show dspfarm profilecommand in privileged EXEC mode.

show dspfarm profile [ profile-identifier ]

Syntax Description

profile ide ntifier (Optional) Number that uniquely identifies a profile.

Range is from 1 to 65535. There is no default.

Command Modes

Privileged EXEC (#)

Command History

Release

12.3(8)T

Modification

This command was introduced.

Usage Guidelines

Use the show dspfarm profile command to verify that the association between Skinny Client Control Protocol

(SCCP) Cisco Unified CallManager and the DSP farm profiles match your organizational plan.

The output of the show dspfarm profile command differs depending on the services configured in the profile.

Examples

The following is sample output from the

show dspfarm profile

command:

Router#

show dspfarm profile

Dspfarm Profile Configuration

Profile ID = 6, Service = TRANSCODING, Resource ID = 1

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP Status : ASSOCIATED

Resource Provider : FLEX_DSPRM Status : UP

Number of Resource Configured : 4

Number of Resource Available : 4

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

Codec : g729br8, Maximum Packetization Period : 60

RSVP : ENABLED

TRP : FW-TRAVERSAL ENABLED

Dspfarm Profile Configuration

Profile ID = 27, Service = CONFERENCING, Resource ID = 2

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Cisco IOS Voice Command Reference - S commands

475

show dial-peer through show gatekeeper zone prefix show dspfarm profile

Application : SCCP Status : ASSOCIATED

Resource Provider : FLEX_DSPRM Status : UP

Number of Resource Configured : 6

Number of Resource Available : 6

Codec Configuration

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Dspfarm Profile Configuration

Profile ID = 34, Service = MTP, Resource ID = 1

Profile Description :

Profile Service Mode : secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP Status : ASSOCIATED

Resource Provider : NONE Status : UP

Number of Resource Configured : 2

Number of Resource Available : 2

Hardware Configured Resources : 1

Hardware Available Resources : 1

Software Resources : 1

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

TRP : FW-TRAVERSAL ENABLED

The table below describes the significant fields shown in the display.

Table 70: show dspfarm profile Field Descriptions

Field

Profile ID

Service

Resource ID

Profile Description

Profile Service Mode

Profile Admin State

Description

Displays the profile ID number.

Displays the service that is associated with the profile.

Displays the ID number that the profile is associated with in the Cisco CallManager register.

Displays the description of the profile.

The status of the profile service. It can be either

Secure or Non Secure.

Displays the status of the profile. If the Profile Admin

State is DOWN, use the no shutdown command in

DSP farm profile configuration mode.

476

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Field

Profile Operation State

Application

Number of Resource Configured

Number of Resource Available

Hardware Configured Resources

Hardware Available Resources

Software Resources

Codec Configuration

RSVP

TRP

show dspfarm profile

Description

Displays the status of the DSP farm profiles registration process with the Cisco CallManager.

Status options are as follows:

• ACTIVE--The profile is registered with the

Cisco Unified CallManager.

• ACTIVE IN PROGRESS--The profile is still registering with the Cisco Unified CallManager.

Wait for the profile to finish registering.

• DOWN--The profile is not registering with the

Cisco Unified CallManager. Check the connectivity between the DSP farm gateway and the Cisco Unified CallManager.

• DOWN IN PROGRESS--The profile is deregistering from the Cisco Unified

CallManager and deallocating the DSP resources.

• RESOURCE ALLOCATED--The DSP resources for this profile are allocated or reserved.

Displays the routing protocol used.

Maximum number of sessions that are supported by a profile.

Total number of resources that are configurable.

Number of sessions configured in the profile.

Number of sessions available for this profile.

Number of software sessions configured for this profile (applicable only to MTP profiles).

Lists the codecs that are configured.

Note

Media Termination Point (MTP) profile supports only one codec per profile.

Resource Reservation Protocol (RSVP) support for this profile.

Displays whether firewall traversal is enabled for

Trusted Relay Point.

Cisco IOS Voice Command Reference - S commands

477

show dspfarm profile

Related Commands

Command dsp services dspfarm dspfarm profile show media resource status show dial-peer through show gatekeeper zone prefix

Description

Configures DSP farm services for a specified voice card.

Enters DSP farm profile configuration mode and defines a profile for DSP farm services.

Displays the current media resource status.

478

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show dsp-group

show dsp-group

To display digital signal processor (DSP) group information including both voice and video information, use the show dsp-group command in user EXEC or privileged EXEC mode.

show dsp-group {all| slot slot-number| video [all| slot slot-number]| voice [all| slot slot-number]}

Syntax Description all slot

slot-number

video voice

Displays DSP information for all DSP group.

Displays DSP information for the specified slot.

Slot used in the DSP group.

Displays information on video resources.

Displays information on voice resources.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

15.1(4)M

Modification

This command was introduced.

Usage Guidelines

The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs), high-density voice (HDV) transcoding/conferencing DSP farms

(NM-HDV-FARMs), or packet voice data module (PVDM) slots to provide DSP resources.

Examples

The following shows sample output from several forms of the show dsp-groupcommand. The fields are self explanatory.

Router#

show dsp-group all

DSP groups on slot 0: dsp 1:

State: UP, firmware: 28.0.103

Max signal/voice channel: 32/32

Max credits: 480, Voice credits: 0, Video credits: 480 num_of_sig_chnls_allocated: 32

Transcoding channels allocated: 0

Group: FLEX_GROUP_VIDEO_POOL, complexity: FLEX

Video Credits Max: 480, Share: 0, Reserved (rounded-up): 480

Video Group: VIDEO_CONF, rsc id: 2, mode: VCONF_HETE

Session: 0, maximum participants: 4

Video Transcoding channels reserved credits: 480

Video Transcoding channels allocated: 1

Encoder: inactive, credit reserved: 480

Slot: 0

Cisco IOS Voice Command Reference - S commands

479

show dial-peer through show gatekeeper zone prefix show dsp-group

Device idx: 0

PVDM Slot: 0

Dsp Type: SP2600 dsp 2:

State: UP, firmware: 28.0.103

Max signal/voice channel: 32/32

Max credits: 480, Voice credits: 0, Video credits: 480 num_of_sig_chnls_allocated: 32

Transcoding channels allocated: 0

Group: FLEX_GROUP_VIDEO_POOL, complexity: FLEX

Video Credits Max: 480, Share: 0, Reserved (rounded-up): 480

Video Group: VIDEO_CONF, rsc id: 2, mode: VCONF_HETE

Session: 0, maximum participants: 4

Video Transcoding channels reserved credits: 480

Video Transcoding channels allocated: 3

Decoder: inactive, credits reserved: 160

Decoder: inactive, credits reserved: 160

Decoder: inactive, credits reserved: 160

Slot: 0

Device idx: 0

PVDM Slot: 0

Dsp Type: SP2600

DSP groups on slot 1:

This command is not applicable to slot 1

DSP groups on slot 2:

This command is not applicable to slot 2

DSP groups on slot 3:

This command is not applicable to slot 3

Related Commands

Command dsp service dspfarm dspfarm (DSP farm) voice service dsp-reservation voice-card

Description

Configures DSP farm services for a specified voice card.

Enables DSP-farm service.

Configures the percentage of DSP resources are reserved for voice services and enables video services to use the remaining DSP resources.

This command is required to enable video services.

Enters voice-card configuration mode.

480

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show echo-cancel

show echo-cancel

To display the echo-cancellation information of T1/E1 multiflex voice/WAN interface cards, use the show

echo-cancel command in privileged EXEC mode.

show echo-cancel hardware status slot-number

Syntax Description hardware status

slot-number

Displays information about the hardware accelerated

EC device.

Displays the allocation status.

The slot number of the interface cards.

Command Modes

Privileged EXEC (#)

Command History

Release

12.4(24)T

Modification

This command was introduced in a release earlier than Cisco IOS Release

12.4(24)T.

Usage Guidelines

Hardware echo cancellation is restricted to the same baseboard voice/WAN interface card (VWIC) on which the daughter card (EC-MFT-32 and EC-MFT-64) is installed and cannot be shared by other T1/E1 controllers.

Examples

The following is sample output from the show echo-cancel hardware statuscommand:

Router#

show echo-cancel hardware status

ECAN CH Assigned DSP ID VOICEPORT EC NLP COV LAW

===============================================================

0 yes 8 1/0/0 on off on u-Law

1

2

3

4

5 no no no no no

-

-

-

-

-

-

-

-

-

off off off on on off on off on on on on u-Law u-Law on u-Law on u-Law on u-Law

The table below describes the significant fields shown in the display.

Table 71: show echo-cancel Field Descriptions

Field

ECAN CH

Assigned

Description

Total channels in the slot.

Status of the assigned channels.

Cisco IOS Voice Command Reference - S commands

481

show echo-cancel

Field

DSP ID

VOICEPORT

EC

NLP

COV

show dial-peer through show gatekeeper zone prefix

Description

Digital Signaling Processor (DSP) identification number for the assigned channels.

Voice port of the channels.

Echo Cancellation status of the assigned channels.

Status of the Non-Linear Processor (NLP).

Echo cancellation Coverage status of the assigned channels.

482

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show event-manager consumers

show event-manager consumers

To display event-manager statistics for debugging purposes, use the show event-manager consumerscommand in privileged EXEC mode.

show event-manager consumers

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release

12.3(4)T

Modification

This command was introduced.

Examples

The following example shows one call (two call legs) going through the gateway:

Router#

show event-manager consumers

Hash table indexed by AAA_UNIQUE_ID

Uid Consumer_id Consumer_hdl evt_type

00000015 0002

00000015 0002

65B35570

65B35570

START

STOP

00000016 0002

00000016 0002

65B34ECC

65B34ECC

START

STOP

The table below lists and describes the significant output fields.

Table 72: show event-manager consumers Field Descriptions

Field

Uid

Consumer_id

Consumer_hdl evt_type

Description

User ID.

ID of the consumer client process.

Handler of the consumer client process.

Event type.

Related Commands

Command show voice statistics csr interval accounting

Description

Displays all accounting CSRs specified by interval number.

Cisco IOS Voice Command Reference - S commands

483

show dial-peer through show gatekeeper zone prefix show event-manager consumers

Command show voice statistics csr interval aggregation show voice statistics csr since-reset accounting show voice statistics csr since-reset aggregation-level show voice statistics csr since-reset all show voice statistics interval-tag show voice statistics memory-usage

Description

Displays signaling CSRs specified by interval number.

Displays all accounting CSRs since the last reset.

Displays all signaling CSRs since the last reset.

Displays all CSRs since the last reset.

Displays the configured interval numbers.

Displays current memory usage.

484

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show frame-relay vofr

show frame-relay vofr

To display information about the FRF.11 subchannels being used on Voice over Frame Relay (VoFR) data link connection identifiers (DLCIs), use the show frame-relay vofr command in privileged EXEC mode.

show frame-relay vofr [interface [dlci [ cid ]]]

Syntax Description

interface dlci cid

(Optional) Specific interface type and number for which you want to display FRF.11 subchannel information.

(Optional) Specific data link connection identifier for which you want to display FRF.11 subchannel information.

(Optional) Specific subchannel for which you want to display information.

Command Default

If this command is entered without a specified interface, FRF.11 subchannel information is displayed for all

VoFR interfaces and DLCIs configured on the router.

Command Modes

Privileged EXEC (#)

Command History

Release

12.0(4)T

Modification

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810 series.

Examples

The following is sample output from this command when an interface is not specified:

Router# show frame-relay vofr interface vofr-type dlci cid cid-type

Serial0/0.1

Serial0/0.1

VoFR

VoFR

16

16

4

5 data call-control

Serial0/0.1

Serial0/1.1

VoFR 16

VoFR cisco 17

10

4 voice data

The following is sample output from this command when an interface is specified:

Router# show frame-relay vofr serial0 interface

Serial0 vofr-type

VoFR dlci

16 cid

4 cid-type data

Serial0

Serial0

VoFR

VoFR

16

16

5

10 call-control voice

Cisco IOS Voice Command Reference - S commands

485

show dial-peer through show gatekeeper zone prefix show frame-relay vofr

The following is sample output from this command when an interface and a DLCI are specified:

Router# show frame-relay vofr serial0 16

VoFR Configuration for interface Serial0 dlci vofr-type cid cid-type

16 VoFR 4 data input-pkts

0

16 VoFR

16 VoFR

5 call-control

10 voice

85982

2172293 output-pkts dropped-pkts

0 0

86099

6370815

0

0

The following is sample output from this command when an interface, a DLCI, and a CID are specified:

Router# show frame-relay vofr serial0 16 10

VoFR Configuration for interface Serial0 dlci 16 vofr-type VoFR cid 10 cid-type voice input-pkts 2172293 output-pkts 6370815 dropped-pkts 0

The table below describes significant fields shown in this output.

Table 73: show frame-relay vofr Field Descriptions

Field

interface vofr-type cid cid-type input-pkts output-pkts dropped-pkts

Description

Number of the interface that has been selected for observation of FRF.11 subchannels.

Type of VoFR DLCI being observed.

Portion of the specified DLCI that is carrying the designated traffic type. A DLCI can be subdivided into 255 subchannels.

Type of traffic carried on this subchannel.

Number of packets received by this subchannel.

Number of packets sent on this subchannel.

Total number of packets discarded by this subchannel.

Related Commands

Command show call active voice show call history voice show dial-peer voice show frame-relay fragment show frame-relay pvc

Description

Displays the contents of the active call table.

Displays the contents of the call history table.

Displays configuration information and call statistics for dial peers.

Displays Frame Relay fragmentation details.

Displays statistics about PVCs for Frame Relay interfaces.

486

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Command show voice-port show frame-relay vofr

Description

Displays configuration information about a specific voice port.

Cisco IOS Voice Command Reference - S commands

487

show dial-peer through show gatekeeper zone prefix show gatekeeper calls

show gatekeeper calls

To display the status of each ongoing call of which a gatekeeper is aware, use the show gatekeeper

callscommand in privilegedEXEC mode.

show gatekeeper calls [history]

Syntax Description history

(Optional) Displays call history information along with internal error codes at the gatekeeper. The number of disconnected calls displayed in response to this command is the number specified in the

call-history max-size number command. Use of this

max-size number helps to reduce CPU usage in the storage and reporting of this information.

Command Default

The default expression of this command displays information for all active calls detected on the gatekeeper.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(2)NA

12.0(3)T

12.0(5)T

12.1(5)XM2

12.2(4)T

12.2(2)XB1

12.2(11)T

12.4(4)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.0(3)T.

The output for this command was changed.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

The history keyword was added to display historical information on disconnected calls.

488

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper calls

Usage Guidelines

Use this command to show all active calls currently being handled by a particular Multimedia Conference

Manager (MCM) gatekeeper. If you force a disconnect for either a particular call or all calls associated with a particular MCM gatekeeper by using the clear h323 gatekeeper call command, the system does not display information about those calls.

Using the history keyword displays the number of disconnected calls specified in the call-history max-size

number command. Use of this max-sizenumber helps to reduce CPU usage in the storage and reporting of this information.

Examples

The following is sample output showing active calls:

Router# show gatekeeper calls

Total number of active calls = 1.

LocalCallID

12-3339

Endpt(s):Alias src EP:epA dst EP:[email protected]

GATEKEEPER CALL INFO

====================

E.164Addr

Age(secs) BW

94 768(Kbps)

CallSignalAddr

10.0.0.0

Port

1720

RASSignalAddr

10.0.0.0

Port

1700 src PX:pxA dst PX:pxB

10.0.0.0

1720 10.0.0.00

24999

255.255.255.0

1720 255.255.255.0

24999

The table below describes the significant fields shown in the display.

Table 74: show gatekeeper calls Field Descriptions

Field

LocalCallID

Age(secs)

BW(Kbps)

Endpt

Alias

E.164Addr

CallSignalAddr

Port

RASSignalAddr

Port

Description

Identification number of the call.

Age of the call, in seconds.

Bandwidth in use, in kilobytes per second.

Role of each endpoint (terminal, gateway, or proxy) in the call (originator, target, or proxy) and the call signaling and Registration, Admission, and Status

(RAS) protocol address.

H.323-Identification (ID) or Email-ID of the endpoint.

E.164 address of the endpoint.

Call-signaling IP address of the endpoint.

Call-signaling port number of the endpoint.

RAS IP address of the endpoint.

RAS port number of the endpoint.

Cisco IOS Voice Command Reference - S commands

489

show gatekeeper calls

Related Commands

Command clear h323 gatekeeper call call history max show dial-peer through show gatekeeper zone prefix

Description

Forces the disconnection of a specific call or of all calls active on a particular gatekeeper.

Specifies the number of records to be kept in the history table.

490

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper circuits

show gatekeeper circuits

To display the circuit information on a gatekeeper, use the show gatekeeper circuits command in privileged

EXEC mode.

show gatekeeper circuits [{begin| exclude| include} expression]

Syntax Description begin exclude include

expression

(Optional) Displays all circuits, beginning with the line containing the expression.

(Optional) Displays all circuits, excluding those containing the expression.

(Optional) Displays all circuits, including those containing the expression.

(Optional) Word or phrase used to determine what lines are displayed.

Command Default

Shows all circuit information.

Command Modes

Privileged EXEC

Command History

Release

12.2(11)T

Modification

This command was introduced.

Usage Guidelines

Use this command to display current configuration information about the circuits that are registered with the gatekeeper.

Examples

The following command displays the circuit information for the gatekeeper:

Router#

show gatekeeper circuits

Circuit Endpoint Max Calls Avail Calls Resources

-------

CarrierA

---------------- ----------- ---------

Total Endpoints: 2

CarrierB

3640-gw1

5400-gw1

25

23

Total Zones: 1

25

19

Available

Unavailable

The table below describes the fields shown in this output.

Zone

----

MsPacmanGK

Cisco IOS Voice Command Reference - S commands

491

show dial-peer through show gatekeeper zone prefix show gatekeeper circuits

Table 75: show gatekeeper circuits Field Descriptions

Field

Circuit

Endpoint

Max Calls

Avail Calls

Resources

Zone

Total Endpoints

Total Zones

Description

Name of the each circuit connected to the gatekeeper.

Name of each H.323 endpoint.

Maximum number of calls that circuit can handle.

Number of new calls that the circuit can handle at the current time.

Whether the circuit’s resources have exceeded the defined threshold limits. The endpoint

resource-threshold command defines these thresholds.

Zone that supports the endpoint. The zone circuit-id command assigns a zone to an endpoint.

Total number of endpoints supported by the circuit.

Total number of zones supported by the circuit.

Related Commands

Command endpoint resource-threshold zone circuit-id

Description

Sets a gateway’s capacity thresholds in the gatekeeper.

Assigns a remote zone to a carrier.

492

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper cluster

show gatekeeper cluster

To display all the configured gatekeeper clusters information, use the show gatekeeper cluster command in user EXEC or privileged EXEC mode.

show gatekeeper cluster

Syntax Description

This command has no arguments or keywords.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release1.25

12.1(5)XM

12.2(2)T

12.2(2)XB1

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(2)T.

This command was integrated into Cisco IOS Release 12.2(2)XB1 and implemented on the Cisco AS5850 router.

Examples

The following is sample output from the show gatekeeper cluster command. Field descriptions are self-explanatory.

Router#

show gatekeeper cluster

CONFIGURED CLUSTERS

Cluster Name

------------

Cluster A

Type

----

===================

Local

Local Zone

----------

AGK1

Elements

--------

AGK2

IP

--

192.168.200.254 1719

Cluster B Remote

AGK3

BGK1

BGK2

BGK3

192.168.200.223 1719

192.168.200.257 1719

192.168.200.258 1719

192.168.200.259 1719

Related Commands

Command show gatekeeper endpoints show gatekeeper performance stats show gatekeeper zone cluster

Description

Displays the status of all registered endpoints for a gatekeeper.

Displays the performance statistics on the the gatekeeper level message.

Displays the dynamic status of all local clusters.

Cisco IOS Voice Command Reference - S commands

493

show dial-peer through show gatekeeper zone prefix show gatekeeper endpoint circuits

show gatekeeper endpoint circuits

To display information on all registered endpoints and carriers or trunk groups for a gatekeeper, use the show

gatekeeper endpoint circuitscommand in privileged EXEC mode.

show gatekeeper endpoint circuits [{begin| exclude| include} expression]

Syntax Description

| begin

| exclude

| include

expression

(Optional) Displays all circuits, beginning with the line that contains expression.

(Optional) Displays all circuits, excluding those that contain expression.

(Optional) Displays all circuits, including those that contain expression.

(Optional) Word or phrase used to determine what lines are displayed.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(2)NA

12.0(5)T

12.2(11)T

Modification

This command was introduced.

The display format was modified for H.323 Version 2.

The display format was modified to show the E.164 ID, carrier and trunk group data, and total number of active calls.

Usage Guidelines

Use this command to display current configuration information about the endpoints and carriers registered with the gatekeeper. Note that you must type the pipe (|) before any of the optional keywords.

Examples

The following command displays the circuit information for the gatekeeper:

Router#

show gatekeeper endpoint circuits

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name

--------------- ----- --------------- ----- ---------

172.18.195.120

1720 172.18.195.120

51059 LavenderGK

E164-ID: 4081234

H323-ID: 3640-gw1

Carrier: CarrierA, Max Calls: 25, Available: 25

Type

----

VOIP-GW

Flags

-----

494

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper endpoint circuits

172.18.197.143

1720 172.18.197.143

57071 LavenderGK

H323-ID: 5400-gw1

Carrier: CarrierB, Max Calls: 23, Available: 19

Carrier: CarrierA, Max Calls: 25, Available: 25

Total number of active registrations = 2

The table below describes the fields shown in this output.

Table 76: show gatekeeper endpoint circuits Fields

Field

CallsignalAddr

Port

RASSignalAddr

Port

Zone Name

Type

Flags

E164-ID

H323-ID

Carrier

Max Calls

Available

VOIP-GW

Description

Call signaling IP address of the endpoint. If the endpoint is also registered with an alias, a list of all aliases registered for that endpoint should be listed on the line below.

Call signaling port number of the endpoint.

RAS IP address of the endpoint.

RAS port number of the endpoint.

Zone name (gatekeeper ID) that this endpoint registered in.

Endpoint type (for example, terminal, gateway, or

MCU).

S--Endpoint is statically entered from the

aliascommand rather than being dynamically registered through RAS messages.

O--Endpoint, which is a gateway, has sent notification that it is nearly out of resources.

E.164 ID of the endpoint.

H.323 ID of the endpoint.

Carrier associated with the endpoint.

Maximum number of calls the circuit can handle.

Number of new calls the circuit can handle currently.

Related Commands

Command endpoint circuit-id h323id endpoint resource-threshold

Description

Assigns a circuit to a non-Cisco endpoint.

Sets a gateway’s capacity thresholds in the gatekeeper.

Cisco IOS Voice Command Reference - S commands

495

show gatekeeper endpoint circuits

Command zone circuit-id show dial-peer through show gatekeeper zone prefix

Description

Assigns a circuit to a remote zone.

496

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper endpoints

show gatekeeper endpoints

To display the status of all registered endpoints for a gatekeeper, use the show gatekeeper endpoints command in privileged EXEC mode.

show gatekeeper endpoints [alternates]

Syntax Description alternates

(Optional) Displays information about alternate endpoints. All information normally included with this command is also displayed.

Command Modes

Privileged EXEC (#)

Command History

Examples

Release

11.3(2)NA

12.0(5)T

12.1(5)XM

12.1(5)XM2

12.2(2)T

12.2(4)T

12.2(2)XB1

12.2(11)T

12.3(1)

Modification

This command was introduced.

The display format was modified for H.323 Version 2.

The alternates keyword was added.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was integrated into Cisco IOS Release 12.2(2)T.

This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T. The registration and call capacity values were added to the output display.

This command was modified to reflect concurrent calls for the endpoints.

The following is sample output from this command:

Router# show gatekeeper endpoints

CallsignalAddr Port RASSignalAddr Port Zone Name Type

-----------------------------------------------

172.21.127.8

1720

H323-ID:[email protected]

172.21.127.8

24999 sj-gk MCU

Voice Capacity Max.=23 Avail.=23

Total number of active registrations = 1

172.21.13.88

1720 172.21.13.88

1719 sj-gk VOIP-GW

F

--

O H323-ID:la-gw

Cisco IOS Voice Command Reference - S commands

497

show dial-peer through show gatekeeper zone prefix show gatekeeper endpoints

The table below describes significant fields shown in this output.

Table 77: show gatekeeper endpoints Field Descriptions

Field

CallsignalAddr

Port

RASSignalAddr

Port

Zone Name

Type

F

Voice Capacity Max.

Avail.

Total number of active registrations

Description

Call signaling IP address of the endpoint. If the endpoint is also registered with an alias (or aliases), a list of all aliases registered for that endpoint should be listed on the line below.

Call signaling port number of the endpoint.

Registration, Admission, and Status (RAS) protocol

IP address of the endpoint.

RAS port number of the endpoint.

Zone name (gatekeeper identification [ID]) to which this endpoint is registered.

Endpoint type (for example, terminal, gateway, or multipoint control unit [MCU]).

S--Endpoint is statically entered from the alias command rather than being dynamically registered through RAS messages. O--Endpoint, which is a gateway, has sent notification that it is nearly out of resources.

Maximum number of channels available on the endpoint.

Current number of channels available on the endpoint.

Total number of endpoints registered with the gatekeeper.

In the following example, the show gatekeeper endpoints output has been modified to reflect concurrent calls for the endpoint. If an endpoint is not reporting capacity and the endpoint max-calls h323id command is not configured, "Voice Capacity Max." and "Avail." will not be shown. "Current.= 2" indicates that the current active calls for the endpoint are 2.

Router# show gatekeeper endpoints

!

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name

--------------- ---------------------------

172.18.200.27

1720 172.18.200.27

49918 GK-1

H323-ID:GW1

Voice Capacity Max.= Avail.= Current.= 2

Type

----

VOIP-GW

Flags

-----

498

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper endpoints

If an endpoint is reporting capacity but the endpoint max-calls h323id command is not configured, "Voice

Capacity Max." and "Avail." will show reported call capacity of the endpoint as follows:

Router# show gatekeeper endpoints

!

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name

----------------------------------------

172.18.200.29

1720 172.18.200.29

53152 GK-2

H323-ID:GW2

Voice Capacity Max.= 23 Avail.= 22 Current.= 1

Type Flags

--------

VOIP-GW

If an endpoint is reporting capacity but the endpoint max-calls h323id command is not configured, "Voice

Capacity Max." will show the maximum calls configured and "Avail." will show the available calls of the endpoint. In this example, "Voice Capacity Max.= 10" is showing that the maximum calls configured for the endpoint are 10. "Avail.= 2" shows that currently available calls for the endpoint are 2. "Current.= 8" shows that current active calls for the endpoint are 8.

Router# show gatekeeper endpoints

!

GATEKEEPER ENDPOINT REGISTRATION

================================

CallSignalAddr Port RASSignalAddr Port Zone Name

----------------------------------------

172.18.200.27

1720 172.18.200.27

49918 GK-1

H323-ID:GW1

Voice Capacity Max.= 10 Avail.= 2 Current.= 8

The table below describes significant fields in the output examples.

Table 78: show gatekeeper endpoints Field Descriptions

Type

----

VOIP-GW

Flags

-----

Field

CallsignalAddr

Port

RASSignalAddr

Port

Zone Name

Type

Flags

Description

Call signaling IP address of the endpoint. If the endpoint is also registered with an alias (or aliases), a list of all aliases registered for that endpoint should be listed on the line below.

Call signaling port number of the endpoint.

Registration, Admission, and Status (RAS) protocol

IP address of the endpoint.

RAS port number of the endpoint.

Zone name (gatekeeper ID) to which this endpoint is registered.

The endpoint type (for example, terminal, gateway, or multipoint control unit [MCU]).

S--Endpoint is statically entered from the alias command rather than being dynamically registered through RAS messages. O--Endpoint, which is a gateway, has sent notification that it is nearly out of resources.

Cisco IOS Voice Command Reference - S commands

499

show dial-peer through show gatekeeper zone prefix show gatekeeper endpoints

Related Commands

Command endpoint resource-threshold show gatekeeper endpoint circuits show gatekeeper gw-type-prefix show gatekeeper zone status show gateway

Description

Sets a gateway’s capacity thresholds in the gatekeeper.

Displays endpoint and carrier or trunk group call capacities.

Displays the gateway technology prefix table.

Displays the status of zones related to a gatekeeper.

Displays the current gateway status.

500

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper gw-type-prefix

show gatekeeper gw-type-prefix

To display the gateway technology prefix table, use the show gatekeeper gw-type-prefix command in privileged EXEC mode.

show gatekeeper gw-type-prefix

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(2)NA

12.0(5)T

12.1(5)XM2

12.2(4)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced.

The display format was modified for H.323 Version 2.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Examples

The following is sample output from this command for a gatekeeper that controls two local zones, sj-gk and la-gk:

Router# show gatekeeper gw-type-prefix

GATEWAY TYPE PREFIX TABLE

===========================

Prefix:12#* (Default gateway-technology)

Zone sj-gk master gateway list:

10.0.0.0:1720 sj-gw1

10.0.0.0:1720 sj-gw2 (out-of-resources)

10.0.0.0:1720 sj-gw3

Zone sj-gk prefix 408....... priority gateway list(s):

Priority 10:

10.0.0.0:1720 sj-gw1

Priority 5:

10.0.0.0:1720 sj-gw2 (out-of-resources)

10.0.0.0:1720 sj-gw3

Prefix:7#* (Hopoff zone la-gk)

Statically-configured gateways (not necessarily currently registered):

10.0.0.0:1720

10.0.0.0:1720

Zone la-gk master gateway list:

10.0.0.0:1720 la-gw1

10.0.0.0:1720 la-gw2

The table below describes significant fields shown in this output.

Cisco IOS Voice Command Reference - S commands

501

show dial-peer through show gatekeeper zone prefix show gatekeeper gw-type-prefix

Table 79: show gatekeeper gw-type-prefix Field Descriptions

Field

Prefix

Zone sj-gk master gateway list

Zone sj-gk prefix 408....... priority gateway list(s)

Priority 10

Priority 5

(out-of-resources)

(Hopoff zone la-gk)

Zone la-gk master gateway list

(Default gateway-technology)

Description

Technology prefix defined with the gw-type-prefix command.

List of all the gateways registered to zone sj-gk with the technology prefix under which they are listed.

(This display shows that gateways sj-gw1, sj-gw2, and sj-gw3 have registered in zone sj-gk with the technology prefix 12#.)

List of prioritized gateways to handle calls to area code 408.

Highest priority level. Gateways listed following

"Priority 10" are given the highest priority when selecting a gateway to service calls to the specified area code. (In this display, gateway sj-gw1 is given the highest priority to handle calls to the 408 area code.)

Any gateway that does not have a priority level assigned to it defaults to priority 5.

Indication that the displayed gateway has sent a

"low-in-resources" notification.

Any call that specifies this technology prefix should be directed to hop off in the la-gk zone, no matter what the area code of the called number is. (In this display, calls that specify technology prefix 7# are always routed to zone la-gk, regardless of the actual zone prefix in the destination address.)

List of all the gateways registered to la-gk with the technology prefix under which they are listed. (This display shows that gateways la-gw1 and la-gw2 have registered in zone la-gk with the technology prefix

7#. No priority lists are displayed here because none were defined for zone la-gk.)

If no gateway-type prefix is specified in a called number, then gateways that register with 12# are the default type to be used for the call.

502

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Field

Statically-configured gateways

Related Commands

Command show gatekeeper calls show gatekeeper endpoints show gateway show gatekeeper gw-type-prefix

Description

List of all IP addresses and port numbers of gateways that are incapable of supplying technology-prefix information when they register. This display shows that, when gateways 1.1.1.1:1720 and 2.2.2.2:1720 register, they are considered to be of type 7#.

Description

Displays the status of each ongoing call of which a gatekeeper is aware.

Displays the status of all registered endpoints for a gatekeeper.

Displays the current gateway status.

Cisco IOS Voice Command Reference - S commands

503

show dial-peer through show gatekeeper zone prefix show gatekeeper performance statistics

show gatekeeper performance statistics

To display performance statistics on the gatekeeper level message, use the show gatekeeper performance

stats command in user EXEC or privileged EXEC mode.

show gatekeeper performance statistics [zone [name zone-name]] [cumulative]

Syntax Description zone

name zone -name

cumulative

(Optional) Displays zone statistics of the gatekeeper.

(Optional) Specifies the zone name or gatekeeper name.

(Optional) Displays the total statistics collected by the gatekeeper since the last reload.

Command Modes

User EXEC (>) Privileged EXEC (#)

Command History

Release

12.1(5)XM

12.2(2)T1

12.2(2)XB1

12.2(15)T

12.4(5)

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(2)T.

This command was implemented on the Cisco AS5850.

This command was modified. The zone, name, and cumulative keywords were added and the zone-name argument was added.

This command was modified. Command output was enhanced to include counters for:

• Automatic rejections (ARJs) sent due to an ARQ access-list denial.

• Location rejections (LRJs) sent due to an LRQ access-list denial.

Usage Guidelines

Use this command to display the statistics on calls, registration, calls routed to other gatekeepers, and calls used via zone processing.

When the cumulative keyword is used along with zone name keywords displays the total statistics for the specified zone, from the starting time of the gatekeeper. These values are not reset when the clear h323

gatekeeper statscommand is used.

This command displays statistical data related to the router. You can identify the number of call initiation events using the following messages:

504

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix

Examples show gatekeeper performance statistics

• Automatic repeat request (ARQ)

• Admission confirmation (ACF)

• Admission rejection (ARJ)

You can identify endpoint contact events that have been requested and either confirmed or rejected on the router using the following:

• Location request (LRQ)

• Location confirm (LCF)

• Location reject (LRJ)

The counts associated with overload and the number of endpoints sent to alternate gatekeepers that are associated with overload conditions are also displayed. Only when the router experiences an overload condition do these counters reveal a value other than zero. The real endpoint count simply displays the number of endpoints registered on this router platform. The time stamp displays the start time when the counters started capturing the data. When you want to request a new start period, enter the clear h323 gatekeeper stats command. The counters are reset and the time stamp is updated with the new time.

You can identify remote gatekeeper contacts that have been requested and either confirmed or rejected on the router using the following messages:

• Location confirm (LCF)

• Location rejection (LRJ)

• Location request (LRQ)

You can identify zone-level or gatekeeper-level registration statistics using the following messages:

• Registration confirmation (RCF)

• Registration rejection (RRJ)

• Registration request (RRQ)

You can identify zone-level or gatekeeper-level unregistration statistics using the following messages:

• Unregistration confirmation (UCF)

• Unregistration rejection (URJ)

• Unregistration request (URQ)

The following is the example of basic output from the show gatekeeper performance stats command. The basic output specifies that the counters are reset using the clear h323 gatekeeper stats command and the output displays the statistcs from the last reset.

Router#

show gatekeeper performance stats

-----Gatekeeper Performance Statistics-----

Performance statistics captured since: 20:09:00 UTC Thu Sep 15 2005

Gatekeeper level Admission Statistics:

ARQs received: 1

ARQs received from originating endpoints: 0

ACFs sent: 1

ACFs sent to the originating endpoint: 0

Cisco IOS Voice Command Reference - S commands

505

show dial-peer through show gatekeeper zone prefix show gatekeeper performance statistics

ARJs sent: 0

ARJs sent to the originating endpoint: 0

ARJs sent due to overload: 0

ARJs sent due to ARQ access-list denial: 0

Number of concurrent calls: 0

Number of concurrent originating calls: 0

Gatekeeper level Location Statistics:

LRQs received: 3

LRQs sent: 0

LCFs received: 0

LCFs sent: 1

LRJs received: 0

LRJs sent: 2

LRJs sent due to overload: 0

LRJs sent due to LRQ access-list denial: 2

Gatekeeper level Registration Statistics:

RRJ due to overload: 0

Total Registered Endpoints: 2

Gatekeeper level Disengage Statistics:

DRQs received: 1

DRQs sent: 0

DCFs received: 0

DCFs sent: 1

DRJs received: 0

DRJs sent: 0

Gatekeeper viazone message counters: inARQ: 0 infwdARQ: 0 inerrARQ: 0 inLRQ: 0 infwdLRQ: 0 inerrLRQ: 0 outLRQ: 0 outfwdLRQ: 0 outerrLRQ: 0 outARQ: 0 outfwdARQ: 0 outerrARQ: 0

Load balancing events: 0

The following is the example of cumulative output from the show gatekeeper performance stats command.

The cumulative output specifies that the counters are not reset and the output displays the total statistcs from the starting time of the gatekeeper.

Router#

show gatekeeper performance stats zone name voip3-2600-2

Performance statistics for zone voip3-2600-2

-----Zone Level Performance Statistics-----

Performance statistics captured since: 00:17:00 UTC Mon Mar 1 1993

Zone level Admission Statistics:

ARQs received: 1

ARQs received from originating endpoints: 0

ACFs sent: 1

ACFs sent to the originating endpoint: 0

ARJs sent: 0

ARJs sent to the originating endpoint: 0

Number of concurrent total calls: 0

Number of concurrent originating calls: 0

Zone level Location Statistics:

LRQs received: 1

LRQs sent: 0

LCFs received: 0

LCFs sent: 1

LRJs received: 0

LRJs sent: 0

Zone level Registration Statistics:

Full RRQs received: 1

Light RRQs received: 574

RCFs sent: 576

RRJs sent: 0

Total Registered Endpoints: 1

Zone level UnRegistration Statistics:

URQs received: 0

506

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper performance statistics

URQs sent: 0

UCFs received: 0

UCFs sent: 0

URJs received: 0

URJs sent: 0

URQs sent due to timeout: 0

Zone level Disengage Statistics:

DRQs received: 1

DRQs sent: 0

DCFs received: 0

DCFs sent: 1

DRJs received: 0

DRJs sent: 0

The table below shows significant fields shown in the displays. Most of the fields are self-explanatory and are not listed the table.

Table 80: show gatekeeper performance statistics Field Descriptions

Field

Full RRQs received

Light RRQs received

Description

A full registration request (RRQ) contains all registration information that is used for successful registration.

A light RRQ contains abbreviated registration information that is used to maintain an existing registration.

Related Commands

Command clear h323 gatekeeper stats

Description

Clears statistics about gatekeeper performance.

Cisco IOS Voice Command Reference - S commands

507

show dial-peer through show gatekeeper zone prefix show gatekeeper servers

show gatekeeper servers

To display a list of currently registered and statically configured triggers on a gatekeeper router, use the show

gatekeeper servers command in EXEC mode.

show gatekeeper servers [ gkid ]

Syntax Description

gkid

(Optional) Local gatekeeper name to which this trigger applies.

Command Modes

EXEC (#)

Command History

Release

12.1(1)T

12.2(2)XB

12.2(8)T

12.2(11)T

12.2(15)T12

12.3(8)T

12.3(9)

Modification

This command was introduced on the Cisco 2500 series, Cisco 2600 series, Cisco

3600 series, Cisco 7200, and Cisco MC3810.

The output of this command was modified to show additional server statistics, including the following: gatekeeper server timeout value; Gatekeeper Transaction

Message Protocol (GKTMP) version installed; number of Registration Request

(RRQ), Registration Response (RRQ), Response Confirmation (RCF), and

Response Reject (RRJ) messages received; timeouts encountered; average response time; and if the server is usable.

This command was integrated into Cisco IOS Release 12.2(8)T.

This command was implemented on the Cisco 3700 series.

The command was modified to show additional server statistics.

The command was modified to show additional server statistics.

The command was modified to show additional server statistics.

Usage Guidelines

Use this command to show all server triggers (whether dynamically registered from the external servers or statically configured from the command-line interface) on this gatekeeper. If the gatekeeper ID is specified, only triggers applied to the specified gatekeeper zone appear. If the gatekeeper ID is not specified, server triggers for all local zones on this gatekeeper appear.

Examples

The following is sample output from this command:

Router#

show gatekeeper servers

508

Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper servers

GATEKEEPER SERVERS STATUS

=========================

Gatekeeper Server listening port: 8250

Gatekeeper Server timeout value: 30 (100ms)

GateKeeper GKTMP version: 4.1

Gatekeeper-ID: Gatekeeper1

------------------------

RRQ Priority: 5

Server-ID: Server43

Server IP address: 209.165.200.254:40118

Server type: dynamically registered

Connection Status: active

Trigger Information:

Trigger unconditionally

Server Statistics:

REQUEST RRQ Sent=0

RESPONSE RRQ Received = 0

RESPONSE RCF Received = 0

RESPONSE RRJ Received = 0

Average response time(ms)=0

Server Usable=TRUE

Timeout Statistics:

Server-ID: Server43

Server IP address: 209.165.200.254:40118

Server type: dynamically registered

Connection Status: active

Timeout Encountered=0

The table below describes significant fields shown in this output.

Table 81: show gatekeeper servers Field Descriptions

Field

GateKeeper GKTMP version

RRQ Priority

Server-ID

Server IP address

Server type

Connection Status

Trigger Information

REQUEST RRQ

RESPONSE RRQ

RESPONSE RCF

RESPONSE RRJ

Description

Version of Gatekeeper Transaction Message Protocol installed.

Registration priority.

Server ID name.

Server IP address.

Type of server.

Whether the connection is active or inactive.

Which Registration, Admission, and Status (RAS) messages the Cisco IOS gatekeeper forwards to the external application.

Registration requests received.

Registration responses received.

Response confirmations received.

Response reject messages received.

Cisco IOS Voice Command Reference - S commands

509

show gatekeeper servers

Related Commands

Command debug gatekeeper server endpoint circuit-id h323id server registration-port server trigger arq show dial-peer through show gatekeeper zone prefix

Description

Traces all the message exchanges between the Cisco

IOS gatekeeper and the external applications.

Tracks call capacity information on the gatekeeper.

Configures a listening port on the gatekeeper for server registration.

Configures static triggers on the gatekeeper.

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Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper status

show gatekeeper status

To display overall gatekeeper status, including authorization and authentication status and zone status, use the show gatekeeper statuscommand in privileged EXEC mode.

show gatekeeper status

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(2)NA

12.0(3)T

12.1(5)XM

12.2(2)T

12.2(2)XB

12.2(8)T

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.0(3)T.

This command was modified to show information about load balancing and vendor-specific attributes.

This command was integrated into Cisco IOS Release 12.2(2)T.

This command was modified to show information about server flow control.

This command was integrated into Cisco IOS Release 12.2(8)T.

Examples

The following is sample output from this command:

Router#

show gatekeeper status

Gatekeeper State: UP

Load Balancing: DISABLED

Flow Control: ENABLED

Zone Name: snet-3660-3

Accounting: DISABLED

Endpoint Throttling: DISABLED

Security: DISABLED

Maximum Remote Bandwidth: unlimited

Current Remote Bandwidth: 0 kbps

Current Remote Bandwidth (w/ Alt GKs): 0 kbps

The table below describes significant fields shown in this output.

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show dial-peer through show gatekeeper zone prefix show gatekeeper status

Table 82: show gatekeeper status Field Descriptions

Field

Gatekeeper State

Load Balancing

Flow Control

Zone Name

Accounting

Endpoint Throttling

Security

Bandwidth

Description

Gatekeeper state has the following values:

• UP is operational.

• DOWN is administratively shut down.

• INACTIVE is administratively enabled; that is, the no shutdown command has been issued, but no local zones have been configured.

• HSRP STANDBY indicates that the gatekeeper is on hot standby and will take over when the currently active gatekeeper fails.

Whether load balancing is enabled.

Whether server flow control is enabled.

Zone name to which the gatekeeper belongs.

Whether authorization and accounting features are enabled.

Whether endpoint throttling is enabled.

Whether security features are enabled.

Maximum remote bandwidth, current remote bandwidth, and current remote bandwidth with alternate gatekeepers.

Related Commands

Command show gatekeeper servers

Description

Displays statistics about the gatekeeper.

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Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper status cluster

show gatekeeper status cluster

To display information about each element of a local cluster, such as the amount of memory used, the number of active calls, and the number of endpoints registered on the element, use the show gatekeeper status cluster command in privileged EXEC mode.

show gatekeeper status cluster

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(5)XM1

12.2(2)T

12.2(2)XB1

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(2)T.

This command was implemented on the Cisco AS5850.

Examples

The following command displays information about elements of a local cluster, two of whose components are RoseGK and LavenderGK:

Router#

show gatekeeper status cluster

CLUSTER INFORMATION

Hostname

--------

%Mem

===================

Active

%CPU Calls

Endpoint Last

Count Announce

RoseGK

LavenderGK

------------

72

30

0

1

1

0

--------

Local Host

4

--------

14s

Related Commands

Command show gatekeeper endpoints show gatekeeper performance statistics show gatekeeper zone cluster

Description

Displays the status of all registered endpoints for a gatekeeper.

Displays information about the number of calls accepted and rejected, and finds the number of endpoints sent to other gatekeepers.

Displays the dynamic status of all local clusters.

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show dial-peer through show gatekeeper zone prefix show gatekeeper zone cluster

show gatekeeper zone cluster

To display the dynamic status of all local clusters, use the show gatekeeper zone clustercommand in privilegedEXEC mode.

show gatekeeper zone cluster

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

12.1(5)XM1

12.2(2)T

12.2(2)XB1

Modification

This command was introduced.

This command was integrated into Cisco IOS Release 12.2(2)T.

This command was implemented on the Cisco AS5850.

Examples

The following command displays information about the current bandwidth values and about when the last announcement message from the alternate gatekeeper was received. In the following example, PRI represents the priority value assigned to an alternate gatekeeper. This field ranges from 0 to 127, with 127 representing the lowest priority.

Router#

show gatekeeper zone cluster

LOCAL CLUSTER INFORMATION¸6t

============================

TOT BW INT BW REM BW LAST ALT GK

LOCAL GK NAME ALT GK NAME PRI (kbps) (kbps) (kbps) ANNOUNCE STATUS

----------------------------------------------------

ParisGK

NiceGK

GenevaGK

ZurichGK

120

100

0

0

0

0

0

0

7s

7s

CONNECTED

CONNECTED

Related Commands

Command

timer cluster -element announce

zone cluster local zone remote

Description

Defines the time interval between successive announcement messages exchanged between elements of a local cluster.

Defines a local grouping of gatekeepers.

Statically specifies a remote zone if DNS is unavailable or undesirable.

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Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper zone prefix

show gatekeeper zone prefix

To display the zone prefix table, use the show gatekeeper zone prefixcommand in privileged EXEC mode.

show gatekeeper zone prefix [all]

Syntax Description all

(Optional) Displays the dynamic zone prefixes registered by each gateway.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(2)NA

12.2(15)T

Modification

This command was introduced.

The all keyword was added.

Usage Guidelines

If the all keyword is not specified, this command displays the static zone prefixes only. Use the include filter with the all keyword to display the prefixes associated with a particular gateway. For example, the show

gatekeeper zone prefix all | include GW1 command displays the dynamic prefixes associated with gateway

GW1.

Examples

The following command displays the zone prefix table for the gatekeeper:

Router# show gatekeeper zone prefix

ZONE PREFIX TABLE

=================

GK-NAME E164-PREFIX

------gk2 gk2 gk2 gk2 gk2

-----------

408*

5551001*

5551002*

5553020*

5553020* gk1 gk2 gk2

555....

719*

919*

The following command displays the zone prefix table, including the dynamic zone prefixes, for the gatekeeper:

Router# show gatekeeper zone prefix all

ZONE PREFIX TABLE

===============================================

GK-NAME

-------

E164-PREFIX

-----------

Dynamic GW-priority

------------------gk2 gk2 gk2

408*

5551001*

5551002*

GW1 /5

GW1 /5 GW2 /10

Cisco IOS Voice Command Reference - S commands

515

show dial-peer through show gatekeeper zone prefix show gatekeeper zone prefix

gk2 gk2 gk1 gk2 gk2

5553020*

5553020*

555....

719*

919*

GW1 /8

GW2 /5

The table below describes significant fields shown in this output.

Table 83: show gatekeeper zone prefix Field Descriptions

Field

GK-NAME

E164-PREFIX

Dynamic GW-priority

Description

Gatekeeper name.

E.164 prefix and a dot that acts as a wildcard for matching each remaining number in the telephone number.

Gateway that serves this E164 prefix.

Gateway priority. A 0 value prevents the gatekeeper from using the gateway for that prefix. Value 10 places the highest priority on the gateway. The default priority value for a dynamic gateway is 5.

Related Commands

Command show gatekeeper zone cluster

Description

Displays the dynamic status of all local clusters.

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Cisco IOS Voice Command Reference - S commands

show dial-peer through show gatekeeper zone prefix show gatekeeper zone status

show gatekeeper zone status

To display the status of zones related to a gatekeeper, use the show gatekeeper zone status command in privileged EXEC mode.

show gatekeeper zone status

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(2)NA

12.0(5)T

12.1(5)XM2

12.2(4)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced.

The display format was modified for H.323 Version 2.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Examples

The following is sample output from thiscommand:

Router#

show gatekeeper zone status

GATEKEEPER ZONES

GK name

================

Domain Name RAS Address

--------------------------sj.xyz.com

xyz.com

10.0.0.0

SUBNET ATTRIBUTES :

All Other Subnets :(Enabled)

PORT FLAGS MAX-BW CUR-BW

(kbps) (kbps)

----

1719

----- ------

LS

PROXY USAGE CONFIGURATION : inbound Calls from germany.xyz.com : to terminals in local zone sj.xyz.com :use proxy

------

0 to gateways in local zone sj.xyz.com

:do not use proxy

Outbound Calls to germany.xyz.com

from terminals in local zone germany.xyz.com :use proxy from gateways in local zone germany.xyz.com

:do not use proxy

Inbound Calls from all other zones : to terminals in local zone sj.xyz.com :use proxy to gateways in local zone sj.xyz.com

:do not use proxy

Outbound Calls to all other zones : from terminals in local zone sj.xyz.com :do not use proxy from gateways in local zone sj.xyz.com

:do not use proxy

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show dial-peer through show gatekeeper zone prefix show gatekeeper zone status

tokyo.xyz.co xyz.com

milan.xyz.co xyz.com

10.0.0.0

10.0.0.0

1719 RS

1719 RS

The table below describes significant fields shown in this output.

Table 84: show gatekeeper zone status Field Descriptions

Field

GK name

Domain Name

RAS Address

FLAGS

Description

0

0

Gatekeeper name (also known as the zone name), which is truncated after 12 characters in the display.

Domain with which the gatekeeper is associated.

Registration, Admission, and Status (RAS) protocol address of the gatekeeper.

Displays the following information:

• S = static (CLI-configured, not

DNS-discovered)

• L = local

• R = remote

MAX-BW

CUR-BW

SUBNET ATTRIBUTES

PROXY USAGE CONFIGURATION

Maximum bandwidth for the zone, in kbps.

Current bandwidth in use, in kbps.

List of subnets controlled by the local gatekeeper.

Inbound and outbound proxy policies as configured for the local gatekeeper (or zone).

Related Commands

Command show gatekeeper calls show gatekeeper endpoints show gateway

Description

Displays the status of each ongoing call of which a gatekeeper is aware.

Displays the status of registered endpoints for a gatekeeper.

Displays the current gateway status.

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Cisco IOS Voice Command Reference - S commands

show gateway through show modem relay statistics

show gateway, page 521

show h323 calls preserved, page 523

show h323 gateway, page 525

show h323 gateway prefixes, page 533

show http client cache, page 535

show http client cache, page 539

show http client cookie, page 542

show http client history, page 544

show http client secure status, page 545

show http client statistics, page 547

show interface dspfarm, page 551

show interfaces cable-modem, page 557

show iua as, page 562

show iua asp, page 565

show media resource status, page 568

show mediacard, page 570

show mgcp, page 573

show mgcp connection, page 583

show mgcp endpoint, page 588

show mgcp nas, page 591

show mgcp profile, page 595

show mgcp srtp, page 601

show mgcp statistics, page 604

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519

show modem relay statistics, page 609

show gateway through show modem relay statistics

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Cisco IOS Voice Command Reference - S commands

show gateway through show modem relay statistics show gateway

show gateway

To display the current status of the gateway, use the show gateway command in privileged EXEC mode.

show gateway

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC (#)

Command History

Release

11.3(6)NA2

12.0(5)T

12.1(5)XM2

12.2(4)T

12.2(2)XB1

12.2(11)T

Modification

This command was introduced.

The display format was modified for H.323 Version 2.

This command was implemented on the Cisco AS5350 and Cisco AS5400.

This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.

This command was implemented on the Cisco AS5850.

This command was integrated into Cisco IOS Release 12.2(11)T.

Examples

The following sample output shows the report that appears when the gateway is not registered with a gatekeeper:

Router# show gateway

Gateway gateway1 is not registered to any gatekeeper

Gateway alias list

H323-ID gateway1

H323 resource thresholding is Enabled but NOT Active

H323 resource threshold values:

DSP: Low threshold 60, High threshold 70

DS0: Low threshold 60, High threshold 70

This following sample output indicates that an E.164 address has been assigned to the gateway:

Router# show gateway

Gateway gateway1 is registered to Gatekeeper gk1

Gateway alias list

E.164 Number 5551212

H323-ID gateway1

The following sample output shows the report that appears when the gateway is registered with a gatekeeper and H.323 resource threshold reporting is enabled with the resource threshold command:

Router# show gateway

Gateway gateway1 is registered to Gatekeeper gk1

Gateway alias list

H323-ID gateway1

Cisco IOS Voice Command Reference - S commands

521

show gateway through show modem relay statistics show gateway

H323 resource thresholding is Enabled and Active

H323 resource threshold values:

DSP: Low threshold 60, High threshold 70

DS0: Low threshold 60, High threshold 70

The following sample output shows the