Configuring voice entities

H3C MSR Router Series
Comware 5 Voice Configuration Guide
New H3C Technologies Co., Ltd.
http://www.h3c.com.hk
Software version: MSR-CMW520-R2516
Document version: 20170612-C-1.12
Copyright © 2006-2017, New H3C Technologies Co., Ltd. and its licensors
All rights reserved
No part of this manual may be reproduced or transmitted in any form or by any means without prior written
consent of New H3C Technologies Co., Ltd.
Trademarks
H3C,
, H3CS, H3CIE, H3CNE, Aolynk,
, H3Care,
, IRF, NetPilot, Netflow, SecEngine,
SecPath, SecCenter, SecBlade, Comware, ITCMM and HUASAN are trademarks of New H3C Technologies
Co., Ltd.
All other trademarks that may be mentioned in this manual are the property of their respective owners
Notice
The information in this document is subject to change without notice. Every effort has been made in the
preparation of this document to ensure accuracy of the contents, but all statements, information, and
recommendations in this document do not constitute the warranty of any kind, express or implied.
Preface
This configuration guide describes fundamentals and configuration of Voice Entity, Voice Subscriber,
Dial Plan, SIP, H.323, Call Services, and so on.
This preface includes the following topics about the documentation:
•
Audience.
•
Conventions
•
Obtaining documentation
•
Technical support
•
Documentation feedback
Audience
This documentation is intended for:
•
Network planners.
•
Field technical support and servicing engineers.
•
Network administrators working with the H3C MSR series routers.
Conventions
The following information describes the conventions used in the documentation.
Command conventions
Convention
Description
Boldface
Bold text represents commands and keywords that you enter literally as shown.
Italic
Italic text represents arguments that you replace with actual values.
[]
Square brackets enclose syntax choices (keywords or arguments) that are optional.
{ x | y | ... }
Braces enclose a set of required syntax choices separated by vertical bars, from which
you select one.
[ x | y | ... ]
Square brackets enclose a set of optional syntax choices separated by vertical bars,
from which you select one or none.
{ x | y | ... } *
Asterisk marked braces enclose a set of required syntax choices separated by vertical
bars, from which you select a minimum of one.
[ x | y | ... ] *
Asterisk marked square brackets enclose optional syntax choices separated by vertical
bars, from which you select one choice, multiple choices, or none.
&<1-n>
The argument or keyword and argument combination before the ampersand (&) sign
can be entered 1 to n times.
#
A line that starts with a pound (#) sign is comments.
GUI conventions
Convention
Description
Boldface
Window names, button names, field names, and menu items are in Boldface. For
example, the New User window opens; click OK.
Convention
Description
>
Multi-level menus are separated by angle brackets. For example, File > Create >
Folder.
Convention
Description
Symbols
WARNING!
An alert that calls attention to important information that if not understood or followed
can result in personal injury.
CAUTION:
An alert that calls attention to important information that if not understood or followed
can result in data loss, data corruption, or damage to hardware or software.
IMPORTANT:
An alert that calls attention to essential information.
NOTE:
TIP:
An alert that contains additional or supplementary information.
An alert that provides helpful information.
Network topology icons
Convention
Description
Represents a generic network device, such as a router, switch, or firewall.
Represents a routing-capable device, such as a router or Layer 3 switch.
Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that
supports Layer 2 forwarding and other Layer 2 features.
Represents an access controller, a unified wired-WLAN module, or the access
controller engine on a unified wired-WLAN switch.
Represents an access point.
T
Wireless terminator unit.
T
Wireless terminator.
Represents a mesh access point.
Represents omnidirectional signals.
Represents directional signals.
Represents a security product, such as a firewall, UTM, multiservice security
gateway, or load balancing device.
Represents a security module, such as a firewall, load balancing, NetStream, SSL
VPN, IPS, or ACG module.
Examples provided in this document
Examples in this document might use devices that differ from your device in hardware model,
configuration, or software version. It is normal that the port numbers, sample output, screenshots,
and other information in the examples differ from what you have on your device.
Obtaining documentation
To access the most up-to-date H3C product documentation, go to the H3C website at
http://www.h3c.com.hk
To obtain information about installation, configuration, and maintenance, click
http://www.h3c.com.hk/Technical_Documents
To obtain software version information such as release notes, click
http://www.h3c.com.hk/Software_Download
Technical support
service@h3c.com
http://www.h3c.com.hk
Documentation feedback
You can e-mail your comments about product documentation to info@h3c.com.
We appreciate your comments.
Contents
Voice overview ················································································1 Introduction to VoIP ···················································································································· 1 VoIP system ······················································································································· 1 Basic VoIP call flow ············································································································· 1 Hardware compatibility with voice ·································································································· 2 VoIP features ····················································································································· 2 Configuring voice functions ·········································································································· 3 Configuration procedure ······································································································· 3 Voice subscriber lines ·········································································································· 5 Voice entities ······················································································································ 5 Voice protocols ··················································································································· 6 Dial plan ···························································································································· 7 Configuring voice entities ···································································8 Overview ·································································································································· 8 Hardware compatibility with voice entities ························································································ 8 Voice entity configuration task list ·································································································· 9 Configuring a POTS voice entity···································································································· 9 POTS voice entity configuration task list ··················································································· 9 Configuration prerequisites ···································································································· 9 Creating a POTS voice entity ································································································· 9 Configuring basic functions ·································································································· 10 Configuring the local POTS voice entity to play ringback tones ···················································· 11 Configuring DTMF transmission ··························································································· 12 Enabling VAD ··················································································································· 12 Configuring options related to dial plan ·················································································· 13 Configuring the jitter buffer ·································································································· 13 Configuring a VoIP voice entity ··································································································· 14 VoIP voice entity configuration task list··················································································· 14 Creating a VoIP voice entity ································································································· 14 Configuring basic functions ·································································································· 15 Configuring DTMF transmission ··························································································· 16 Configuring fast connection and tunneling··············································································· 16 Configuring out-of-band DTMF transmission in fast connection mode ··········································· 17 Configuring out-of-band DTMF transmission with tunneling enabled ············································· 18 Enabling VAD ··················································································································· 18 Configuring options related to dial plan ·················································································· 19 Configuring the jitter buffer ·································································································· 19 Setting the keepalive interval ······························································································· 20 Configuring the timeout interval for RTP streams ············································································ 20 Enabling local call identification ··································································································· 20 Introduction ······················································································································ 20 Configuration procedure ····································································································· 21 Configuring voice call performance-related parameters ···································································· 21 Configuration prerequisites ·································································································· 21 Configuration procedure ····································································································· 21 Configuring global default parameters for voice entities ···································································· 22 Enabling FXO monitoring ··········································································································· 23 Enabling the trap function ·········································································································· 24 Displaying and maintaining voice entity configuration ······································································· 24 Voice entity configuration examples ····························································································· 25 Voice entity configuration example for establishing a VoIP call ···················································· 25 Fast connection ················································································································ 27 Troubleshooting voice entity configuration ····················································································· 28 Busy tone heard immediately after number dialed ···································································· 28 i
Configuring analog voice subscriber lines ············································ 29 Signal tone ····························································································································· 29 FXS voice subscriber line ·········································································································· 29 FXS interface ··················································································································· 29 CID ································································································································ 29 FXO voice subscriber line ·········································································································· 30 FXO interface ··················································································································· 30 CID ································································································································ 30 Busy tone detection ··········································································································· 30 E&M subscriber line·················································································································· 31 E&M interface ··················································································································· 31 E&M start mode ················································································································ 31 Hardware compatibility with analog voice subscriber lines································································· 32 Configuration task list················································································································ 33 Configuring call progress tones ··································································································· 33 Configuration prerequisites ·································································································· 33 Specifying the call progress tones of a country ········································································ 33 Customizing call progress tones for a country ·········································································· 34 Configuring basic functions ········································································································ 34 Configuration prerequisites ·································································································· 34 Configuration procedure ····································································································· 34 Configuring FXS voice subscriber line ·························································································· 35 Configuration prerequisites ·································································································· 35 Configuration guidelines ····································································································· 35 Configuring CID ················································································································ 35 Configuring packet loss compensation mode ··········································································· 36 Setting the electrical impedance ··························································································· 36 Configuring the sending of LCFO signals ················································································ 36 Configuring FXO voice subscriber line ·························································································· 37 Configuration prerequisites ·································································································· 37 Configuration guidelines ····································································································· 37 Configuration procedure ····································································································· 37 Configuring busy tone detection ··························································································· 38 Configuring the off-hook mode ····························································································· 40 Setting ring detection parameters ························································································· 41 Configuring other functions ·································································································· 41 Binding an FXS voice subscriber line to an FXO voice subscriber line ················································· 41 Configuration prerequisites ·································································································· 42 Configuration procedure ····································································································· 42 Configuring E&M voice subscriber line ·························································································· 42 Configuration prerequisites ·································································································· 42 Configuring cable type ········································································································ 43 Configuring signal type ······································································································· 43 Configuring start mode ······································································································· 43 Enabling E&M non-signaling mode ························································································ 44 Enabling E&M analog control signals pass-through ··································································· 45 Configuring analog E&M line failure tone ················································································ 45 Configuring output gain of SLIC chip ····················································································· 46 Configuring DTMF ···················································································································· 46 Introduction to DTMF ········································································································· 46 Configuring DTMF properties ······························································································· 47 Configuring DTMF detection ································································································ 47 Configuring options related to dial plan ························································································· 48 Configuring adjustment functions································································································· 48 Configuration task list ········································································································· 48 Configuring echo cancellation ······························································································ 48 Configuring gain adjustment function ····················································································· 50 Configuring time adjustment function ····················································································· 51 Configuring comfortable noise function··················································································· 51 Configuring PCM pass-through function ················································································· 52 Rebooting a voice card·············································································································· 52 ii
Configuration guidelines ····································································································· 52 Configuration procedure ····································································································· 53 Configuring global default parameters for voice subscriber lines························································· 53 Mirroring PCM, RTP, or voice command data on an analog voice subscriber line ·································· 53 Displaying and maintaining analog voice subscriber lines ································································· 54 Analog voice subscriber line configuration examples ······································································· 54 Configuration example for the FXO voice subscriber line ··························································· 54 Configuration example for one-to-one binding between FXS and FXO ·········································· 55 Troubleshooting analog voice subscriber line configuration ······························································· 57 Failed to hang up ·············································································································· 57 Configuring digital voice subscriber lines ············································· 59 Introduction to E1 and T1··········································································································· 59 Overview ························································································································· 59 E1 and T1 voice functions ··································································································· 59 E1 and T1 interfaces ·········································································································· 59 Features of E1 and T1 ········································································································ 60 Hardware compatibility with digital voice subscriber lines ·································································· 61 E1 and T1 configuration task list·································································································· 61 Configuring basic parameters for an E1 voice interface ···································································· 62 Configuring a TDM clock source ··························································································· 62 Configuring the framing format and line coding format ······························································· 63 Creating a TS set ·············································································································· 64 Set the physical state change suppression interval on an E1 interface ·········································· 64 Restoring default settings for an E1 voice interface ··································································· 64 Configuring basic parameters for a T1 voice interface ······································································ 64 Configuring a TDM clock source ··························································································· 65 Configuring the framing format and line coding format ······························································· 65 Creating a TS set ·············································································································· 65 Set a physical state change suppression interval on a T1 interface ·············································· 66 Restoring default settings for a T1 voice interface····································································· 66 Configuring the voice subscriber line for a TS set ············································································ 66 Configuration prerequisites ·································································································· 66 Configuring basic functions for the voice subscriber line ···························································· 66 Configuring the DTMF detection sensitivity ············································································· 67 Configuring the volume adjustment function ············································································ 68 Configuring the echo adjustment function ··············································································· 68 Configuring the comfortable noise function·············································································· 69 Configuring options related to dial plan ·················································································· 69 Binding logical voice subscriber line to POTS entity ········································································· 69 Configuring R2 signaling············································································································ 70 Introduction to R2 signaling ································································································· 70 Configuring basic R2 signaling parameters ············································································· 75 Configuring R2 digital line signaling ······················································································· 78 Configuring R2 interregister signaling ···················································································· 79 Configuring PRI ······················································································································· 80 Configuring DSS1 and QSIG signaling ··················································································· 80 Enabling the transmission of QSIG signaling over a SIP network ················································· 81 Configuring digital E&M signaling ································································································ 81 Configuring a start mode ····································································································· 81 Enabling E&M non-signaling mode ························································································ 83 Configuring receive and transmit signaling ·············································································· 83 Configuring the time adjustment function ················································································ 84 Querying the circuits of a timeslot or a range of timeslots ··························································· 85 Configuring digital LGS signaling ································································································· 85 Configuration the time adjustment function·············································································· 85 Querying the circuits of a timeslot or a range of timeslots ··························································· 85 Configuring a BSV BRI interface ································································································· 86 BSV interface ··················································································································· 86 Configuration prerequisites ·································································································· 86 Configuration procedure ····································································································· 86 Configuring AMD ····················································································································· 87 iii
Enabling the AMD function ·································································································· 87 Configuring AMD parameters ······························································································· 87 Configuring reverse charging function ·························································································· 88 Enabling the router to treat DISCONNECT messages with PI 8 as standard DISCONNECT messages ······ 89 Mirroring PCM, RTP packets, or voice command data on a digital voice subscriber line ·························· 89 Displaying and maintaining digital voice subscriber lines ·································································· 90 Digital voice subscriber line configuration examples ········································································ 90 E1 R2 signaling and digital E&M signaling configuration example ················································ 90 E1 voice DSS1 signaling configuration example ······································································· 92 QSIG tunneling configuration example ··················································································· 95 Troubleshooting digital voice subscriber line configuration ································································ 97 Failure of call connection from router to PSTN ········································································· 97 Configuring dial plans ······································································ 98 Overview ································································································································ 98 Dial plan process··············································································································· 98 Regular expression ············································································································ 99 Introduction to number substitution ······················································································ 100 Hardware compatibility with dial plans ························································································ 101 Configuration task list·············································································································· 101 Configuring calling numbers permitted to call out ·········································································· 101 Configuring call authority control ······························································································· 102 Configuring match templates for a subscriber group································································ 102 Binding a subscriber group to a voice entity ·········································································· 102 Enabling private line auto ring-down ··························································································· 103 Configuring a number match mode ···························································································· 103 Configuration prerequisites ································································································ 103 Configuring a global number match mode ············································································· 103 Configuring a dial terminator ······························································································ 104 Number match mode configuration example·········································································· 104 Configuring match order of voice entity selection rules ··································································· 106 Configuration prerequisites ································································································ 106 Configuration procedure ··································································································· 106 Configuration example of voice entity selection priority rules ····················································· 107 Configuring voice entity type selection priority rules ································································ 108 Configuration example of voice entity type selection priority rules ·············································· 108 Configuring the voice entity search function ·········································································· 110 Configuration example of the voice entity search function ························································ 110 Configuring a number priority peer ····························································································· 113 Configuration prerequisites ································································································ 113 Configuration procedure ··································································································· 113 Configuring a maximum-call-connection set ················································································· 113 Configuration prerequisites ································································································ 113 Configuration procedure ··································································································· 113 Configuring number substitution ································································································ 114 Configuration prerequisites ································································································ 114 Configuration procedure ··································································································· 114 Configuring a number sending mode ·························································································· 116 Configuration prerequisites ································································································ 117 Configuration procedure ··································································································· 117 Configuring a dial prefix ··········································································································· 117 Configuration prerequisites ································································································ 117 Configuration procedure ··································································································· 117 Displaying and maintaining dial plan configuration ········································································ 118 Dial plan configuration examples ······························································································· 118 Configuring number substitution ························································································· 118 Configuring the match order for voice entity selection ······························································ 121 Configuring the maximum-call-connection set ········································································ 123 Configuring call authority control ························································································· 125 Configuring SIP ··········································································· 128 Overview ······························································································································ 128 iv
Terminology ··················································································································· 128 SIP functions and features ································································································ 129 SIP messages ················································································································ 130 SIP fundamentals ············································································································ 130 Support for transport layer protocols ·························································································· 133 SIP security ·························································································································· 133 Signaling encryption ········································································································· 133 Media flow encryption ······································································································· 134 TLS-SRTP combinations ··································································································· 135 Support for basic QSIG call ······································································································ 135 VRF-aware SIP ····················································································································· 136 Hardware compatibility with SIP ································································································ 136 SIP configuration task list········································································································· 136 Configuring SIP UA registration································································································· 137 Configuration prerequisites ································································································ 138 Configuring SIP authentication information············································································ 138 Configuring registrar information for a SIP UA ······································································· 139 Configuring proxy server information for a SIP UA ·································································· 139 Configuring registration timers ···························································································· 140 Configuring call failure-triggered re-registration ······································································ 141 Configuring fuzzy telephone number registration ···································································· 141 Enabling SIP registration function ······················································································· 141 Configuring SIP server keepalive and backup ·············································································· 142 Configuring SIP routing ··········································································································· 142 Configuration prerequisites ································································································ 143 Configuration procedure (1: destination IP address) ································································ 143 Configuration procedure (2: SIP proxy server) ······································································· 143 Configuration procedure (3: destination domain) ···································································· 143 Configuration procedure (4: ENUM) ···················································································· 144 Configuring user information····································································································· 145 Configuring outbound SIP proxy server information for a SIP UA······················································ 145 Configuring transport layer protocol for SIP calls ··········································································· 145 Configuring UDP or TCP for outgoing SIP calls ······································································ 145 Configuring UDP or TCP for incoming SIP calls ····································································· 147 Configuring SIP security ·········································································································· 147 Configuring TLS for SIP sessions ······················································································· 147 Configuring media flow protocols for SIP calls ······································································· 149 Specifying the URL scheme for outgoing SIP calls ········································································ 149 Configuring SIP extensions ······································································································ 150 Strict SIP routing ············································································································· 150 Configuring out-of-band SIP DTMF transmission mode ··························································· 151 Configuring source IP address binding for SIP messages ························································ 151 Configuring a domain name for the SIP UA ··········································································· 152 Configuring SIP compatibility ····························································································· 153 Configuring user-agent and server header fields ···································································· 154 Configuring SIP extensions for caller identity and privacy ························································· 155 Configuring call release cause code mapping ········································································ 156 Configuring periodic refresh of SIP sessions ········································································· 156 Enabling early media negotiation ························································································ 157 Configuring VRF-aware SIP ····································································································· 158 Displaying and maintaining SIP UAs ·························································································· 158 SIP UA configuration examples ································································································· 159 Configuring direct calling for SIP UAs ·················································································· 159 Configuring proxy server involved calling for SIP UAs ····························································· 160 Configuring DNS involved calling for SIP UAs········································································ 162 Configuring out-of-band SIP DTMF transmission mode ··························································· 163 Configuring SIP extensions for caller identity and privacy ························································· 164 Configuring TCP to carry outgoing SIP calls ·········································································· 165 Configuring TLS to carry outgoing SIP calls ·········································································· 167 Configuring SIPS URL scheme for outgoing SIP calls ····························································· 170 Configuring SRTP for SIP calls ··························································································· 171 Troubleshooting ····················································································································· 173 v
Failed to set up calls in the proxy server approach to SIP routing ··············································· 173 Failed to register with the registrar ······················································································ 173 Failed to set up point-to-point calls ······················································································ 173 Failed to send register requests ·························································································· 173 Failed to set up point-to-point SIP calls over TLS ··································································· 174 Configuring SIP local survival ························································· 175 Hardware compatibility with SIP local survival ·············································································· 175 Configuration task list·············································································································· 176 Configuring an operation mode for the local SIP server ·································································· 176 Configuring user information····································································································· 178 Specifying a trusted node ········································································································ 178 Configuring call authority control ······························································································· 179 Configuring a call rule ······································································································· 179 Applying a call rule set ······································································································ 179 Configuring an area prefix ········································································································ 180 Configuring a call route ··········································································································· 180 Displaying and maintaining the SIP local survival feature configuration ·············································· 181 SIP local survival feature configuration examples ·········································································· 181 Configuring the local SIP server to operate in the alone mode ··················································· 181 Configuring the local SIP server to operate in the alive mode (method 1) ···································· 183 Configuring the local SIP server to operate in the alive mode (method 2) ···································· 185 Configuring the call authority control ···················································································· 187 Configuring an area prefix ································································································· 189 Configuring a call route ····································································································· 191 Configuring SIP trunk ···································································· 193 Background ·························································································································· 193 Features ······························································································································ 194 Typical applications ················································································································ 194 Protocols and standards ·········································································································· 195 Hardware compatibility with SIP trunk ························································································· 195 SIP trunk configuration task list ································································································· 195 Enabling the SIP trunk function ································································································· 196 Configuring a SIP server group ································································································· 196 Creating a SIP server group ······························································································ 196 Enabling the real-time switching function ·············································································· 197 Configuring the keepalive and redundancy functions ······························································· 198 Configuring source address binding ···················································································· 199 Configuring a SIP trunk account ································································································ 200 Configuring a SIP trunk account for registration ····································································· 200 Configuring registration timers for a SIP trunk account ···························································· 201 Configuring call routes for outbound calls ···················································································· 201 Binding a SIP server group to the VoIP voice entity ································································ 201 Specifying the destination address ······················································································ 202 Specifying the proxy server used for outbound calls ································································ 202 Configuring call match rules ······························································································· 203 Configuring call routes for inbound calls ······················································································ 204 Enabling codec transparent transfer ··························································································· 204 Enabling media flow-around ····································································································· 205 Enabling delayed offer to early offer conversion ············································································ 205 Enabling codec transcoding ····································································································· 206 Enabling address hiding ·········································································································· 206 Enabling call forwarding ·········································································································· 207 Enabling call transfer ·············································································································· 207 Enabling midcall signaling pass-through ····················································································· 207 Displaying and maintaining SIP trunk configuration ······································································· 208 SIP trunk configuration examples ······························································································ 208 Configuring a SIP server group with only one member server ··················································· 208 Configuring a SIP server group with multiple member servers ··················································· 211 Configuring call match rules ······························································································· 212 vi
Configuring H.323 ········································································ 214 H.323 architecture ·················································································································· 215 H.323 fundamentals ··············································································································· 216 Gatekeeper discovery ······································································································ 216 Registration···················································································································· 216 Address translation ·········································································································· 216 Admission control ············································································································ 216 Call setup ······················································································································ 218 Call proceeding ··············································································································· 218 Alerting ························································································································· 218 Connection ···················································································································· 218 Capability negotiation ······································································································· 218 Opening/closing logical channels ························································································ 218 Complete release ············································································································ 218 Disconnection ················································································································· 218 Hardware compatibility with H.323 ····························································································· 219 H.323 gateway configuration task list ························································································· 219 Configuring basic H.323 gateway functions ··········································································· 220 Configuring registration password ······················································································· 221 Enabling security calling ··································································································· 221 Displaying and maintaining the H.323 gateway ············································································· 222 H.323 gateway configuration example ························································································ 222 Troubleshooting H.323 gateway ································································································ 224 Registration failure··········································································································· 224 Configuring call services ································································ 225 Call waiting ···················································································································· 225 Call hold ························································································································ 225 Call forwarding················································································································ 225 Call transfer ··················································································································· 225 Call backup ···················································································································· 226 Hunt group ····················································································································· 226 Call barring ···················································································································· 226 Message waiting indication ································································································ 226 Three-party conference ···································································································· 226 Silent monitor and barge in services ···················································································· 226 Calling party control ········································································································· 227 Door opening control ········································································································ 227 Support for SIP voice service of VCX ··················································································· 227 Hardware compatibility with call services ····················································································· 227 Call services configuration task list ···························································································· 228 Configuring call waiting ··········································································································· 228 Configuration prerequisites ································································································ 229 Enabling and disabling call waiting by using keys ··································································· 229 Configuring call waiting by using command lines ···································································· 229 Configuration example ······································································································ 230 Configuring call hold ··············································································································· 230 Configuration prerequisites ································································································ 230 Enabling call hold using command lines ··············································································· 230 Configuring the tone playing mode for call hold ······································································ 230 Configuration example ······································································································ 231 Configuring call forwarding ······································································································· 231 Configuration prerequisites ································································································ 231 Enabling and disabling call forwarding by using keys ······························································ 231 Configuring call forwarding by using command lines ······························································· 232 Configuration example ······································································································ 233 Configuring call transfer··········································································································· 234 Configuration prerequisites ································································································ 234 Configuring call transfer by using command lines ··································································· 234 Configuration example ······································································································ 234 Configuring call backup ··········································································································· 235 vii
Configuring hunt group ············································································································ 235 Configuration prerequisites ································································································ 235 Enabling hunt group ········································································································· 235 Configuring hunt group priority level ···················································································· 236 Configuration example ······································································································ 236 Configuring incoming call barring······························································································· 237 Configuration prerequisites ································································································ 237 Enabling and disabling incoming call barring by using keys ······················································ 237 Enabling incoming call barring by using command lines ··························································· 237 Configuration example ······································································································ 237 Configuring outgoing call barring ······························································································· 238 Configuration prerequisites ································································································ 238 Enabling and disabling outgoing call barring by using keys ······················································· 238 Enabling outgoing call barring by using command lines ··························································· 238 Configuration example ······································································································ 238 Configuring MWI ···················································································································· 239 Configuration prerequisites ································································································ 239 Enabling and disabling MWI ······························································································ 239 Specifying the voice mailbox server ····················································································· 239 Displaying and maintaining MWI ························································································· 239 Configuring three-party conference ···························································································· 240 Configuration prerequisites ································································································ 240 Enabling three-party conference by using keys ······································································ 240 Enabling three-party conference by using command lines ························································ 240 Configuration example ······································································································ 241 Configuring silent monitor and barge in ······················································································· 241 Configuration prerequisites ································································································ 241 Configuring three-party conference in active participation mode by using keys ····························· 241 Configuring three-party conference in active participation mode by using command lines ··············· 242 Configuring calling party control ································································································ 242 Configuration prerequisites ································································································ 242 Configuring calling party control ·························································································· 242 Configuring door opening control ······························································································· 242 Configuration prerequisites ································································································ 242 Configuring door opening control ························································································ 242 Configuring Feature service ····································································································· 243 Configuration prerequisites ································································································ 243 Enabling and disabling Feature service setting by using keys ··················································· 243 Configuring Feature service by using command lines ······························································ 245 Configuration example ······································································································ 246 Configuring a number priority peer ····························································································· 246 Call services configuration examples ·························································································· 246 Call waiting ···················································································································· 247 Call forwarding busy ········································································································ 248 Call transfer ··················································································································· 249 Hunt group ····················································································································· 251 MWI ····························································································································· 252 Three-party conference ···································································································· 255 Silent monitor and barge in ································································································ 257 Configuring call watch ··································································· 262 Overview ······························································································································ 262 Call watch concepts················································································································ 262 Call watch group ············································································································· 262 Monitoring rule ················································································································ 262 Call watch mode ············································································································· 263 Hardware compatibility with call watch ························································································ 263 Configuring call watch for an E1/T1 interface ··············································································· 263 Configuring a call watch group ··························································································· 263 Associating the E1/T1 interface with the call watch group ························································· 264 Displaying and maintaining call watch ························································································ 264 Call-watch configuration examples····························································································· 264 viii
Monitoring local interfaces ································································································· 264 Monitoring remote IP addresses ························································································· 265 Configuring fax over IP ·································································· 268 FoIP protocols and standards ··································································································· 268 Fax flow ······························································································································· 268 Hardware compatibility with FoIP······························································································· 269 FoIP configuration task list ······································································································· 269 Configuring fax interworking protocol ··················································································· 270 Enabling CNG fax switchover ····························································································· 272 Enabling ECM for fax ······································································································· 272 Configuring fax faculty transmission mode ············································································ 273 Configuring maximum fax rate ···························································································· 273 Configuring fax training mode ···························································································· 274 Configuring threshold of local training ·················································································· 275 Configuring transmit energy level of gateway carrier ······························································· 276 Configuring T.38 faculty description compatibility ··································································· 276 Configuring global default parameters for fax········································································· 277 Displaying and maintaining FoIP configuration ············································································· 278 FoIP configuration examples ···································································································· 278 Configuring FoIP ············································································································· 278 Configuring SIP modem pass-through·················································································· 279 Configuring customizable IVR ························································· 281 Overview ······························································································································ 281 Advantages ·························································································································· 281 Customizable voice prompts ······························································································ 281 Various codecs ··············································································································· 281 Flexible node configuration ································································································ 281 Customizable process ······································································································ 281 Successive jumping ········································································································· 282 Error processing methods ································································································· 282 Timeout processing methods ····························································································· 282 Various types of secondary calls ························································································· 282 Hardware compatibility with customizable IVR ·············································································· 282 Customizable IVR configuration task list ····················································································· 283 Configuring an IVR voice entity ································································································· 283 Creating an IVR voice entity ······························································································ 283 Configuring an IVR voice entity ·························································································· 283 Specifying the ID for a media resource ······················································································· 285 Configuring IVR processing methods globally ·············································································· 286 Creating an IVR node ············································································································· 286 Configuring a Call node ···································································································· 287 Configuring a Jump node ·································································································· 288 Configuring a Service node ······························································································· 289 Displaying and maintaining customizable IVR ·············································································· 289 Customizable IVR configuration examples ··················································································· 290 Call node configuration example 1: dial terminator match, normal secondary call ·························· 290 Call node configuration example 2: number length match, normal secondary call ·························· 292 Call node configuration example 3: number match, normal secondary call ··································· 293 Call node configuration example 4: extension secondary call ···················································· 294 Jump node configuration example ······················································································· 295 Service node configuration example 1·················································································· 297 Service node configuration example 2·················································································· 299 Configuration example for three types of nodes ····································································· 300 Troubleshooting IVR ··············································································································· 303 Invalid node ··················································································································· 303 Loopback node ··············································································································· 303 Node depth exceeds eight levels ························································································ 304 Matching mistake 1 ·········································································································· 305 Matching mistake 2 ·········································································································· 305 ix
Configuring VoFR ········································································· 307 Overview ······························································································································ 307 Fundamental VoFR architecture ························································································· 307 Protocols and standards ··································································································· 307 Call flow in dynamic mode ································································································· 308 Call flow in FRF.11 trunk mode ·························································································· 308 Hardware compatibility with VoFR ····························································································· 309 VoFR configuration task list ······································································································ 309 Configuring VoFR entity ·········································································································· 310 Creating VoFR entity ········································································································ 310 Configuring basic functions ································································································ 310 Configuring DTMF transmission ························································································· 311 Enabling VAD ················································································································· 311 Configuring VoFR voice bandwidth ···························································································· 311 Configuring dynamic mode······································································································· 312 Configuring Huawei-compatible mode ·················································································· 313 Configuring nonstandard-compatible mode ··········································································· 314 Configuring FRF.11 trunk mode ································································································ 315 Configuration prerequisites ································································································ 315 Configuring call mode ······································································································· 315 Configuring PSTN-dialed number ······················································································· 315 Configuring call control protocol ·························································································· 316 Configuring trunk timer length in FRF.11 trunk mode ······························································· 316 Configuring VoFR packets to carry sequence number ····························································· 317 Displaying and maintaining VoFR ······························································································ 317 VoFR configuration examples ··································································································· 317 Huawei-compatible VoFR ·································································································· 317 Nonstandard-compatible VoFR ·························································································· 319 FRF.11 trunk ·················································································································· 320 Concurrent transmission of voice and data············································································ 321 Troubleshooting VoFR ············································································································ 323 Call failure in Huawei-compatible mode ················································································ 323 Poor VoFR quality ··········································································································· 324 Configuring voice RADIUS ····························································· 325 Voice RADIUS call setup process ······························································································ 325 RADIUS provided by voice gateway ··························································································· 326 AAA for voice calls ··········································································································· 326 Voice dialing process ······································································································· 327 Voice prompt ·················································································································· 327 Recording and querying detailed voice call information ···························································· 328 Hardware compatibility with voice RADIUS ·················································································· 328 Voice RADIUS configuration task list ·························································································· 328 Configuring voice RADIUS ······································································································· 330 Configuring accounting method ·························································································· 330 Enabling the accounting function for one-stage dialing users ···················································· 330 Enabling authentication function for one-stage dialing users ····················································· 331 Enabling authorization function for one-stage dialing users ······················································· 331 Configuring rule for saving CDRs ························································································ 332 Configuring an access number ··························································································· 333 Configuring two-stage dialing process ·················································································· 333 Enabling accounting function for two-stage dialing users ·························································· 334 Enabling the authentication function for two-stage dialing users ················································ 334 Enabling the authorization function for two-stage dialing users ·················································· 335 Configuring the method of collecting the digits of called number ················································ 336 Configuring the timeout interval between two digits for two-stage dialing users ····························· 336 Configuring the number of digits in a card number/password ···················································· 337 Configuring the number of redial attempts············································································· 338 Enabling the language selection function ·············································································· 338 Displaying and maintaining voice RADIUS ·················································································· 339 Voice RADIUS configuration example ························································································ 339 x
Card number/password process configuration ······································································· 339 Troubleshooting voice RADIUS ································································································· 342 Index ························································································· 343 xi
Voice overview
Introduction to VoIP
Voice over IP (VoIP) enables IP networks to provide voice services such as plain old telephone
service (POTS). In VoIP, the voice gateway encapsulates voice signals into packets to transmit. IP
telephony is a typical VoIP application.
Interworking between PSTN and IP is implemented through VoIP gateways. VoIP meets the
commercial requirements for PC-to-telephone, telephone-to-PC, and telephone-to-telephone
technologies.
H.323 and Session Initiation Protocol (SIP) are two common protocols used in VoIP. For information
about H.323 and SIP, see "Voice protocols."
VoIP system
For POTS, all functions from the call originator to the call receiver are implemented by the public
switched telephone network (PSTN). VoIP functions differently from POTS, as described in this
section.
Figure 1 VoIP system
In Figure 1, the VoIP gateway provides interfaces for communication between the IP network and
PSTN/ISDN. Users connect to the originating VoIP gateway through PSTN. The originating VoIP
gateway converts analog signals into digital signals and compresses them into voice packets that
can be transmitted over the IP network. The IP network transmits the voice packets to the terminating
VoIP gateway, which converts the voice packets back into recognizable analog signals and then
transmits them to the receiver. This is a complete telephone-to-telephone communication process. In
practice, a gatekeeper (GK) server or SIP server can be applied in the VoIP system to implement
functions such as routing and access control.
Basic VoIP call flow
The following describes a basic VoIP call flow:
1.
A user picks up a telephone, and then the modular voice card detects the user’s off-hook action
in real time.
2.
The modular voice card transmits the off-hook signal to the VoIP signal processing module on
the VoIP gateway.
3.
The VoIP signal processing module generates dial tones.
4.
The user hears dial tones played by the session application and begins dialing before the dial
tone timer expires.
5.
The session application collects the digits dialed by the user.
1
6.
The session application compares the collected digits with the match template while collecting
digits.
7.
After finding a match template for the called number, the originating VoIP gateway maps the
number to the terminating VoIP gateway.
8.
The originating VoIP gateway initiates a VoIP call to the terminating VoIP gateway over the IP
network and establishes a logical channel for the call to send and receive voice data.
9.
The terminating VoIP gateway receives the call from the IP network and seeks the destination
telephone according to the match template. If the call is to be processed by a PBX, the
terminating VoIP gateway passes the call through PSTN signaling to the PBX for processing
until the destination telephone is connected. When the calling party or the called party hangs up,
the conversation ends.
Hardware compatibility with voice
Voice is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
VoIP features
•
Silence compression
To reduce the amount of voice traffic to be transmitted, VoIP can automatically detect the time
ranges of silence in a conversation, stop generating voice traffic, and send a small number of
silence packets within these time ranges.
•
Comfort noise
Silent gaps during a call can be filled by comfortable background noise.
•
QoS
As voice services are highly time-sensitive, the priority transmission of voice packets must be
guaranteed. Some measures such as PQ, CQ, WFQ, CBQ, and RTP can be adopted on the
sender side for this purpose. To ensure an adequate bandwidth for voice transmission, adopt
the CAR mechanism to implement traffic classification and policing.
•
Fax over IP
Based on VoIP, the FoIP system is responsible for setting up of fax channels and receiving and
sending fax data. FoIP implementation involves modulation and demodulation, fax protocol
processing, and IP channel maintenance.
•
One-stage dialing and two-stage dialing
One-stage dialing and two-stage dialing are methods used to connect the calling user to the
called number. If a PBX sends the called number to the VoIP gateway, the VoIP adopts the
one-stage dialing to connect the calling user. If the PBX does not send the called number to the
VoIP gateway, the VoIP gateway adopts the two-stage dialing and plays prompt tones to guide
the calling user to enter information such as a called number.
2
•
Automatic busy-tone detection
Different PBXs are likely to play different busy tones with different frequency spectra. Therefore,
it is hard to recognize a specific busy-tone feature. With the smart busy tone identification
technology, the VoIP gateway samples, calculates, and analyzes the busy tones played by the
PBX to distinguish matching tones. Implement busy-tone detection by configuring these
parameters on interfaces.
Configuring voice functions
Figure 2 shows that voice function configuration includes four parts: voice subscriber line, voice
entity, voice protocol, and dial plan.
Figure 2 Voice function configuration
Voice function
configuration
Voice
subscriber line
Voice
entity
Voice
protocol
Dial plan
Configuration procedure
Figure 3 shows the voice function configuration procedure of the router. For more information,
see Table 1.
3
Figure 3 Voice function configuration procedure
Start
Configure a link
connection
Is the link
available?
No
Yes
Configure voice entity
Is number
substitution
necessary?
Yes
Configure number
substitution for dial
plans
No
Configure voice
subscriber line
Configure number
application for dial plans
Configure voice
protocol
Troubleshoot
No
Is the call
established?
Yes
End
Table 1 Description of the voice function configuration procedure
Operation
Reference
1.
Connect the physical devices according to the network diagram.
N/A
2.
Configure links and routes and make sure the links and routes are
available.
Layer 3—IP Routing
Configuration Guide
3.
Configure voice entities.
Configuring voice entities
Determine whether number substitution is necessary:
If so, configure number substitution for the dial plan.
{
If not, proceed with the steps below.
Configuring dial plans
4.
{
5.
6.
Configure basic parameters for related voice subscriber lines.
The physical characteristics of voice subscriber lines are usually set
to the default.
Configure number application for the dial plan adopted in the network
4
•
•
Configuring analog voice
subscriber lines
Configuring digital voice
subscriber lines
Configuring dial plans
Operation
Reference
diagram.
7.
8.
Configure the following voice protocols according to the service and
networking environment:
{
H.323 protocol
{
SIP protocol
{
Fax protocol
•
•
•
Check whether the network requirements are met:
If so, the configuration is completed.
{
If not, check the fault and perform re-configuration.
N/A
{
Configuring H.323
Configuring SIP
Configuring fax over IP
Voice subscriber lines
Voice subscriber lines, which are connected to telephone network devices such as analog telephone
and PBX, implement all physical layer functions between VoIP gateways and PSTN devices. These
functions include power supply to analog telephones, off-hook state detection, ringing signal
generation, receiving & sending of analog or digital voice calls, and receiving & sending of dialed
digits for call routing.
For more information about the voice subscriber line, see "Configuring analog voice subscriber lines"
and "Configuring digital voice subscriber lines." The router provides the following voice subscriber
lines:
•
FXS analog voice subscriber line, corresponding to an FXS interface. FXS analog voice
subscriber lines are usually connected to FXO subscriber line terminals, such as ordinary
analog telephones, to provide ringing current, ringing voltage, and dial tone.
•
FXO analog voice subscriber line, corresponding to an FXO interface or 2-port loop trunk
interface. FXO analog voice subscriber lines are usually connected to analog telephone
interfaces of PSTN central offices (PBXs).
•
E&M analog voice subscriber line, corresponding to an E&M interface. E&M analog voice
subscriber lines support analog E&M signaling and divide each voice connection into a trunk
circuit side and a signaling unit side (similar to the relationship between DCE and DTE). PBXs
send signals to routers through M lines and receive signals from routers through E lines.
•
Digital E1/T1 voice subscriber line, that is, a TS set or PRI group created on a VE1/VT1
interface card. After a TS set or PRI group and signaling types, such as R2 signaling, digital
E&M signaling, or digital LGS are configured on VE1/VT1 voice interface cards, the system will
automatically generate the corresponding voice subscriber line for the TS set or PRI group. If a
TS set is created, the E1/T1 interface supports R2, digital E&M, and digital LGS signaling. If a
PRI group is created, the E1/T1 interface supports ISDN, where DSS1 and QSIG are commonly
used protocol types.
•
BSV voice subscriber line, which supports ISDN. Generally, a BSV interface is used to connect
an ISDN digital telephone, or used as a trunk interface to connect a PBX digital trunk.
Voice entities
For information about voice entity configuration, see "Configuring voice entities", "Configuring VoFR",
and "Configuring customizable IVR."
There are four kinds of voice entities: POTS entity, VoIP entity, VoFR entity, and IVR entity.
•
A POTS entity corresponds to the local telephone (or PSTN) side. POTS entity configuration
associates a voice subscriber line on the VoIP gateway with a local telephone. The POTS entity
configuration also implements the binding between telephone numbers and voice subscriber
lines.
5
•
A VoIP entity relates a call entity with a routing policy. Compared with the POTS entity, the VoIP
entity corresponds to the IP network side. VoIP configuration implements the binding between
telephone numbers and destination addresses (IP addresses or server addresses).
•
A VoFR entity is used to transmit voice data over a frame relay network.
•
An IVR entity is used to set a customizable interactive voice response system.
Voice protocols
The VoIP gateway can transfer voice or fax over the IP network by using different protocols. The
basic voice protocols that routers support are H.323 and SIP. The fax protocol is T.38.
1.
H.323
H.323 is a standard protocol established by ITU-T. The H.323 protocol stack, implemented at
the application layer, mainly describes terminals, devices, and services used for multimedia
communication without QoS guarantee over an IP network. An H.323 network usually consists
of VoIP gateway, an optional GK, an MCU, and terminals. According to the ITU-T specifications,
the GK should provide H.323 terminals, a gateway, or MCU in LANs or WANs with the following
functions:
{
Address translation
{
Access permission
{
Bandwidth control and management
{
Area management and security check
{
Call control signaling and call management
{
Routing control and accounting
The GK not only controls the call service, but also functions as the central control point within its
management area. The GK implements the control function by exchanging information with the
VoIP gateway. If there is any GK, the router will be under the control of the GK. To implement
the control function of the GK, perform related configurations on the router. For more
information, see "Configuring H.323."
2.
SIP
SIP is the core protocol of the IETF multimedia data and control architecture. SIP is used for
signaling control and communication with a softswitch platform in the IP network. A SIP network
consists of a user agent (SIP endpoint), proxy server, registration server, location server, and
redirect server. Here, the proxy server, registration server, location server, and redirect server
are functional entities. In practice, multiple functional entities can be integrated into one
physical entity.
{
{
{
In a complete SIP system, all SIP endpoints serve as user agents and should register with
the registration server to inform of their locations, session capabilities, and call policies. The
registration server then sends the registration information to the location server for storage.
SIP endpoints use the proxy server to set up calls. SIP endpoints send signaling messages
to the proxy server, and then the proxy server forwards them to the next hop. In this process,
multiple proxy servers might be involved. Eventually, channels are established to transfer
the upper layer voice service.
Unlike the proxy server, the SIP redirect server will not forward the received session request
messages. Instead it will inform the originating SIP endpoints of the addresses of the
terminating SIP endpoints by returning reply messages. The originating SIP endpoints
directly re-originate a session request message to the terminating SIP endpoints. The
terminating SIP endpoints also directly return a reply message to the originating SIP
endpoints.
As a SIP endpoint, the voice router needs to exchange information with the servers to
accomplish functions such as registration. For more information, see "Configuring SIP."
3.
Fax protocol
6
FoIP complies with ITU-T T.30 and T.4 on PSTN and T.38 on the IP network.
{
{
{
T.30 defines the procedures necessary for document transmission between facsimile
terminals on PSTN. It gives detailed descriptions and stipulations on the communication
process, signal format, control signaling, and error correction of Group 3 facsimile terminals
on the general switched telephone network.
T.4 is a standard protocol used for document transmission between Group 3 facsimile
terminals. It standardizes image coding, signaling modulation, rate, transmission time, error
correction, and document transmission of Group 3 facsimile terminals.
T.38 describes the technical features necessary to transfer facsimile document in real time
between Group 3 facsimile terminals over the Internet or other networks by using IP
protocols. It gives descriptions and stipulations on communication mode, message format,
error correction, and part of communication processes.
Before applying the fax service, configure the technical protocols and physical characteristics.
For more information, see "Configuring fax over IP."
Dial plan
Dial plan configuration provides diversified number management functions. Dial plan configuration
involves number substitution and number application.
•
Number substitution means applying the number substitution rules to the calling and called
numbers to substitute them. Number substitution includes number substitution rules and
binding of number substitution rules.
•
Number application means matching numbers, controlling the sending of numbers, and
selecting voice entities according to match templates. Number application includes number
match policy, rules in the match order for voice entity selection, maximum-call-connection set,
number sending mode, and call authority control.
The dial plan configuration directly affects the selection of voice entity and the final call connection.
The dial plan configuration involves global configuration, voice subscriber line configuration, and
voice entity configuration. You can select one or more configurations for a dial plan. The global
configuration acts on calls of the whole VoIP gateway, the voice entity configuration on those of the
voice entity, and the voice subscriber line configuration on those of the voice subscriber line. For
more information, see "Configuring dial plans."
7
Configuring voice entities
Overview
The voice entity configuration involves:
•
POTS voice entity configuration
•
VoIP voice entity configuration
According to the position of the caller or callee, a complete telephone-to-telephone connection can
be divided into four call segments, each of which corresponds to a voice entity.
Figure 4 Two types of voice entities in the VoIP voice communication
Figure 4 shows that two types of voice entities are involved in VoIP communication:
•
POTS voice entity—This type of voice entity corresponds to the local telephone or PSTN side.
POTS voice entity configuration is required to establish connections between physical voice
subscriber-lines and local telephone devices.
•
VoIP voice entity—This type of entity maps telephone numbers with destination addresses.
Contrasted to POTS voice entity, it resides at the IP side. You can configure a VoIP voice entity
to use the SIP or H.323 protocol to make VoIP calls.
Besides the POTS and VoIP voice entities, the device also supports VoFR entity and IVR entity. A
VoFR entity is used to transmit voice data over a frame relay network, and an IVR entity is used to set
a customizable interactive voice response system. For more information about these two types of
voice entities, see "Configuring VoFR" and "Configuring customizable IVR."
Hardware compatibility with voice entities
Voice entities are not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
8
•
MSR3600-51F.
Voice entity configuration task list
Task
Remarks
Configuring a POTS voice entity
Required.
Configuring a VoIP voice entity
Required.
Configuring the timeout interval for RTP streams
Optional.
Enabling local call identification
Optional.
Configuring voice call performance-related parameters
Optional.
Configuring global default parameters for voice entities
Optional.
Enabling the trap function
Optional.
Configuring a POTS voice entity
This section covers the procedures for creating and configuring a POTS voice entity.
POTS voice entity configuration task list
Task
Remarks
Creating a POTS voice entity
Required.
Configuring basic functions
Required.
Configuring the local POTS voice entity to play ringback tones
Optional.
Configuring DTMF transmission
Optional.
Enabling VAD
Optional.
Configuring options related to dial plan
Optional.
Configuring the jitter buffer
Optional.
Configuration prerequisites
Log in to the router equipped with a voice card (for example, an FXS interface card), and enter user
view.
Creating a POTS voice entity
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Create a POTS voice entity and
enter POTS voice entity view.
entity entity-number pots
N/A
9
Step
5.
6.
Command
Remarks
Bind a number template: bind a
number template to a local
subscriber line, or if the POTS
voice entity serves as a trunk,
configure a number template for
the terminating side.
match-template match-string
By default, no number template
is bound to the local voice
subscriber line. If the POTS
voice entity serves as a trunk,
no number template is
configured for the terminating
side.
Bind the local voice subscriber line
to the POTS voice entity.
line line-number
By default, no voice subscriber
line is bound to the POTS voice
entity.
Optional.
7.
8.
Enable the VoIP gateway to
register numbers of the POTS
voice entity with the H.323
gatekeeper or SIP server.
register-number
Set the DSCP value in the ToS
field in the IP packets that carry
the RTP stream of the voice entity.
dscp media dscp-value
By default, after configured with
GK or SIP registration related
parameters, a POTS voice
entity initiates registration to the
H.323 gatekeeper or SIP
server.
Optional.
EF (101110) by default.
Configuring basic functions
After creating a POTS voice entity, you need to configure basic functions of the POTS voice entity.
Configuration guidelines
When you configure basic functions, follow these guidelines:
•
Two communication parties can communicate correctly only if they share some identical codec
algorithms. Therefore, when you configure the compression command, make sure that the
devices on both sides share identical codec algorithms. Otherwise, the call will fail.
•
Support for the codec algorithms varies with cards and the part related to the upper layer (voice
entity) is independent of hardware. Therefore, the upper layer only verifies the maximum range
for each codec. It is up to the DSP to determine whether the value within this range is valid
according to the hardware type. If yes, the upper layer sends the value. If not, the default value
for the codec is adopted. If you find that a configured packetization period does not take effect,
first check whether the value configured for the card and codec is valid.
Configuration procedure
To configure basic functions of the POTS voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS voice entity
and enter POTS voice entity
view.
entity entity-number pots
N/A
Specify the codecs and their
priority levels for the POTS
voice entity.
compression { 1st-level |
2nd-level | 3rd-level | 4th-level }
{ g711alaw | g711ulaw | g723r53
| g723r63 | g726r16 | g726r24 |
Optional.
5.
10
By default, the codec with the first
priority is g729r8, that with the
Step
Command
Remarks
g726r32 | g726r40 | g729a |
g729br8 | g729r8 }
second priority is g711alaw, that
with the third priority is g711ulaw,
and that with the fourth priority is
g723r53.
Only the following interface cards
support the g726 codec:
•
1-port, 2-port, and 4-port
FXS interface cards.
•
1-port, 2-port, and 4-port
FXO interface cards.
•
2-port and 4-port E&M
interface cards.
Optional.
6.
Configure the voice
packetization period for
different codecs.
payload-size { g711 | g723 |
g726r16 | g726r24 | g726r32 |
g726r40 | g729 } time-length
7.
Configure voice entity
description.
description string
Change the management
state of the voice entity from
up to down.
shutdown
8.
The default is 20 milliseconds for
a G.711 codec, and 30
milliseconds for G.723, G.726,
and G.729 codecs.
Optional.
No description by default.
Optional.
By default, the management state
of the voice entity is up.
Configuring the local POTS voice entity to play ringback
tones
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS voice entity
and enter POTS voice entity
view.
entity entity-number pots
N/A
Bind a number template to
the local voice subscriber
line.
match-template match-string
By default, no number template is
bound.
Bind the POTS voice entity to
a voice subscriber line.
line line-number
By default, the POTS voice entity
is not bound to any voice
subscriber line.
5.
6.
Optional.
7.
Enable the local POTS voice
entity to play ringback tones.
send-ring
11
By default, the local POTS voice
entity does not play ringback
tones.
This command is only applicable
for POTS entities bound to a
non-FXS or non-FXO voice
subscriber lines.
Configuring DTMF transmission
In conversation, DTMF digits can be transmitted transparently between originating and terminating
gateways in inband or out-of-band mode:
•
Inband transmission—DTMF dialing tones are transmitted to the peer end as normal voice
packets.
•
Out-of-band transmission—Corresponding information is extracted from DTMF digits and is
then encapsulated in H.245, H.225, SIP, or RTP packets (RFC 2883) for transmission. The
transmission of DTMF digits in RTP packets is also called as NTE mode.
Configuration guidelines
When you configure DTMF transmission, follow these guidelines:
•
Both H.323 and SIP support NTE.
•
Because the implementations of different vendors can differ, H3C recommends that you
configure command outband nte and the same payload type value on both the originating and
terminating sides. Otherwise, NTE negotiation might fail and thus the user cannot receive
DTMF tones.
•
It is forbidden to set the NTE payload type field to 98, which has already been used to identify
nonstandard T.38 fax packets.
•
When the device is connected with other manufacturers' routers for communication, you cannot
set the payload type field to any forbidden by these routers. Otherwise, an NTE negotiation
failure might occur.
Configuration procedure
To configure out-of-band DTMF transmission:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS voice entity
and enter POTS voice entity
view.
entity entity-number pots
N/A
5.
Configure out-of-band DTMF
transmission.
outband { h225 | h245 | nte | sip }
6.
Configure the value of the
payload type used by NTE.
rtp payload-type nte value
Optional.
By default, inband transmission is
adopted.
Optional.
The default is 101.
Enabling VAD
The voice activity detection (VAD) discriminates between silence and speech on a voice connection
according to their energies. VAD reduces the bandwidth requirements of a voice connection by not
generating traffic during periods of silence in an active voice connection. Speech signals are
generated and transmitted only when an active voice segment is detected. Researches show that
VAD can save the transmission bandwidth by 50%.
To enable VAD:
12
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS voice entity
and enter POTS voice entity
view.
entity entity-number pots
N/A
Enable VAD.
vad-on [ g711 | g723r53 |
g723r63 | g729a | g729r8 ]
5.
Disabled by default.
The G.726 codec does not
support VAD.
Configuring options related to dial plan
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS voice entity
and enter POTS voice entity
view.
entity entity-number
pots
N/A
Configure the calling
number permitted to
originate calls to the local
voice entity.
caller-permit
calling-string
5.
Optional.
By default, no calling number is configured,
that is, incoming calls are not restricted.
Optional.
6.
Configure the priority of the
POTS voice entity.
priority priority-order
7.
Configure a dial prefix.
dial-prefix string
8.
Configure the number
sending mode.
send-number
{ digit-number | all |
truncate }
By default, the priority level is 0.
The smaller the number, the higher the
priority.
Optional.
By default, no dial prefix is configured.
Optional.
By default, the truncate mode is used, that
is, numbers matching the wildcard dot (.) at
the end will be sent.
For more information about the above commands, see "Configuring dial plans."
Configuring the jitter buffer
Jitter, packet loss, and packet disorder occurs during the transmission of voice packets over IP
networks. By using a series of adaptive algorithms, the jitter buffer stores the received IP packets for
a period of time and sends the packets in evenly spaced intervals. The jitter buffer reduces packet
delay and jitter, and improves the communication quality.
To configure the jitter buffer:
13
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS voice entity
and enter POTS voice entity
view.
entity entity-number pots
N/A
Optional.
5.
Set the operating mode of
the jitter buffer to adaptive.
jitter-buffer mode adaptive
6.
Set the operating parameters
for the jitter buffer that
operates in the adaptive
mode.
jitter-buffer delay { initial
milliseconds | maximum
milliseconds }
By default, the adaptive mode is
disabled and the jitter buffer does
not buffer any voice packet.
Optional.
By default, the initial buffering
time is 30 milliseconds and the
maximum buffering time is 160
milliseconds.
Configuring a VoIP voice entity
This section covers the procedures for creating and configuring a VoIP voice entity.
VoIP voice entity configuration task list
Task
Remarks
Creating a VoIP voice entity
Required.
Configuring basic functions
Required.
Configuring DTMF transmission
Optional.
Configuring fast connection and tunneling
Optional.
Configuring out-of-band DTMF transmission in fast connection mode
Optional.
Configuring out-of-band DTMF transmission with tunneling enabled
Optional.
Enabling VAD
Optional.
Configuring options related to dial plan
Optional.
Configuring the jitter buffer
Optional.
Setting the keepalive interval
Optional.
Creating a VoIP voice entity
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
14
Step
Command
Remarks
4.
Create VoIP voice entity and
enter its view.
entity entity-number voip
N/A
5.
Configure a number template
on the terminating router.
match-template match-string
By default, no called number
template is configured for the
voice entity.
•
6.
Configure a policy for routing
from the VoIP voice entity to
the terminating VoIP
gateway.
•
address sip { dns
domain-name [ port
port-number ] | enum-group
group-number | ip ip-address
[ port port-number ] | proxy |
server-group
group-number }
address { ip ip-address |
ras }
Use either command.
By default, no policy is configured.
Optional.
7.
Configure an area ID.
area-id string
By default, no area ID is
configured.
If the routing policy is SIP, this
command is unavailable.
8.
Configure the DSCP subfield
of the ToS field in IP packets
of the RTP stream carried by
the VoIP voice entity.
dscp media dscp-value
Optional.
The default is EF (101110).
Configuring basic functions
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a VoIP voice entity
and enter VoIP voice entity
view.
entity entity-number voip
N/A
Optional.
5.
6.
Specify the codecs and their
priority levels for the POTS
voice entity.
compression { 1st-level |
2nd-level | 3rd-level | 4th-level }
{ g711alaw | g711ulaw | g723r53
| g723r63 | g726r16 | g726r24 |
g726r32 | g726r40 | g729a |
g729br8 | g729r8 }
Configure the voice
packetization period for
payload-size { g711 | g723 |
g726r16 | g726r24 | g726r32 |
15
By default, the codec with the first
priority is g729r8, that with the
second priority is g711alaw, that
with the third priority is g711ulaw,
and that with the fourth priority is
g723r53.
Only the following interface cards
support the g726 codec:
•
1-port, 2-port, and 4-port
FXS interface cards.
•
1-port, 2-port, and 4-port
FXO interface cards.
•
2-port and 4-port E&M
interface cards.
Optional.
The default is 20 milliseconds for
Step
7.
Command
Remarks
different codecs.
g726r40 | g729 } time-length
a G.711 codec, and 30
milliseconds for G.723, G.726,
and G.729 codecs.
Configure voice entity
description.
description string
Optional.
No description is configured by
default.
Configuring DTMF transmission
For precautions, see "Configuring DTMF transmission."
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a VoIP voice entity
and enter VoIP voice entity
view.
entity entity-number voip
N/A
5.
Configure out-of-band DTMF
transmission mode.
outband { h225 | h245 |nte |sip }
6.
Configure the value of the
payload type used by NTE.
rtp payload-type nte value
Optional.
By default, the inband DTMF
transmission is configured.
Optional.
The default is 101.
Configuring fast connection and tunneling
Fast connection and tunneling
•
According to the specification of H.225.0 recommendation, fast connection means that a Setup,
CallProceeding, Alerting, or Connect message carries a standard H.245 message (for example,
Open Logical Channel message) so that an RTP/RTCP voice channel can be established
before the gateway (GW) receives a Connect message, avoiding H.245 message exchange on
TCP connection, and thereby shortening connection time. There is no capability negotiation
process in fast connection mode, so the capability of both parties is determined by the
terminating GW. When fast connection is enabled, a Setup message sent from the originating
GW carries locally supported codecs. After receiving this message, the terminating GW selects
one suitable codec to notify the originating GW through a CallProceeding, Alerting, or Connect
message. In this way, both parties adopt this codec for communication.
•
Tunneling means that in fast connection mode, non-standard H.245 messages (for example,
transparent transmission capability of DTMF digit) are encapsulated in an H.225.0 Facility
message to complete capability negotiation and call forwarding. This makes it unnecessary to
establish an independent H.245 TCP connection for transmission of H.245 message.
Enabling fast connection and tunneling on the originating GW
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
16
Step
Command
Remarks
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity
view.
entity entity-number voip
N/A
5.
Enable fast connection.
fast-connect
Disabled by default.
Optional.
In the fast connection mode, the remote
end plays ringback tones by default.
6.
Enable the local end to
play ringback tones.
This command is available on an H.323
voice entity only after fast connection is
enabled for it.
send-ring
If you want the local end to play ringback
tones, carry out this command.
Optional.
7.
Enable tunneling.
Disabled by default.
tunnel-on
This command is applicable only after
the fast connection is enabled.
Enabling fast connection and tunneling on the terminating GW
You can enable or disable fast connection for each outgoing call on the originating GW. If fast
connection is enabled on the originating GW, the terminating GW will determine whether to enable
fast connection for call initialization or enable tunneling, depending on the configurations of the voip
called-start and voip called-tunnel enable commands.
To enable fast connection and tunneling on the terminating GW:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Configure the call
initialization method for the
terminating GW.
voip called-start { fast | normal }
Enable tunneling on the
terminating GW.
voip called-tunnel enable
5.
Optional.
Fast connection by default.
Optional.
Enabled by default.
Configuring out-of-band DTMF transmission in fast
connection mode
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
17
Step
Command
Remarks
5.
Enable fast connection.
fast-connect
Disabled by default.
6.
Configure out-of-band DTMF
transmission.
outband h225
Optional.
Inband DTMF transmission by default.
NOTE:
In actual configuration, to implement the transparent transmission of DTMF digit, you need to
perform configurations for the VoIP voice entity on the originating GW and the POTS voice entity on
the terminating GW. The out-of-band transmission mode of the POTS voice entity should be
identical with that of the VoIP voice entity.
Configuring out-of-band DTMF transmission with tunneling
enabled
In fast connection mode, DTMF is transmitted in H.245 UserInput messages through the tunneling
function, and the transparent transmission of DTMF cannot be completed if either the originating side
or terminating side does not support the tunneling function.
To configure the out-of-band DTMF transmission with tunneling enabled in the fast connection mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
5.
Enable fast connection.
fast-connect
Disabled by default.
Disabled by default.
6.
Enable the tunneling.
tunnel-on
7.
Configure the out-of-band
DTMF transmission.
outband { h225 | h245 }
This command is applicable only
after the fast connection is
enabled.
Optional.
Inband transmission by default.
Enabling VAD
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a VoIP voice entity
and enter VoIP voice entity
view.
entity entity-number voip
N/A
Enable VAD.
vad-on
5.
Optional.
Disabled by default.
18
Step
Command
Remarks
Note that the G.711 codec does
not support VAD.
Configuring options related to dial plan
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a VoIP voice entity
and enter VoIP voice entity
view.
entity entity-number voip
N/A
5.
Configure calling numbers
permitted to originate calls to
the local voice entity.
Optional.
caller-permit calling-string
By default, no calling number is
configured, that is, incoming calls
are not restricted.
Optional.
6.
Configure the priority of the
VoIP voice entity.
priority priority-order
By default, the priority level is 0.
The smaller the number, the
higher the priority.
For more information about the above commands, see "Configuring dial plans."
Configuring the jitter buffer
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a VoIP voice entity
and enter VoIP voice entity
view.
entity entity-number voip
N/A
Optional.
5.
Set the operating mode of
the jitter buffer to adaptive.
jitter-buffer mode adaptive
6.
Set the operating parameters
for the jitter buffer that
operates in the adaptive
mode.
jitter-buffer delay { initial
milliseconds | maximum
milliseconds }
By default, the adaptive mode is
disabled and the voice packets
are not buffered.
Optional.
19
By default, the initial buffering
time is 30 milliseconds and the
maximum buffering time is 160
milliseconds.
Setting the keepalive interval
With the keepalive function enabled, if the next destination of the VoIP voice entity is reachable, the
VoIP voice entity is set to available. Otherwise, the VoIP voice entity is set to unavailable.
If the keepalive function is disabled, the system considers the VoIP voice entity is available by
default.
To set the keepalive interval:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a VoIP voice entity
and enter VoIP voice entity
view.
entity entity-number voip
N/A
Set the keepalive interval.
keepalive [ interval seconds ]
Optional.
5.
By default, keepalive function is
disabled.
Configuring the timeout interval for RTP streams
If the device does not receive any RTP packets during the specified timeout interval, it will disconnect
the established IP call connections.
To configure the timeout interval for RTP streams:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure the timeout
interval for RTP streams.
rtp-detect timeout value
Optional.
By default, the timeout interval for
RTP streams is 120 seconds.
Enabling local call identification
Introduction
As shown in Figure 6, Telephone A originates a call to Telephone B through a SIP server. If the link
between the SIP server and Router B fails, the SIP server forwards the call back to Router A, which
then forwards the call through its FXO interface.
In this case, the call from Telephone A eventually uses two DSP resources. This can be avoided by
enabling the local call identification function, which helps to identify local calls. With this function
enabled in this case, the voice interface corresponding to Telephone A can be directly connected
with the FXO interface, thus saving the DSP resources.
20
NOTE:
• Whether or not DSP resources can be saved depends on the voice card.
• This function can identify local calls originated by SIP, and cannot identify local calls originated by
H.323.
Figure 5 Network diagram
Configuration procedure
To enable local call identification:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enable local call identification.
distinguish-localtalk
Optional.
Not enabled by default.
Configuring voice call performance-related
parameters
This section describes the configuration procedure for voice call performance-related parameters.
Configuration prerequisites
You have completed the required configurations for a voice entity.
Configuration procedure
To configure voice performance-related parameters:
Step
Command
Remarks
1.
system-view
N/A
Enter system view.
21
Step
Command
Remarks
2.
voice-setup
N/A
Enter voice view.
Optional.
3.
4.
5.
Set the duration of
monitoring DSP
buffered data.
vqa dsp-monitor
buffer-time time
Set the DSCP subfield
in the ToS field in the IP
packets that carry the
RTP stream or voice
signaling.
vqa dscp { media | signal }
dscp-value
Set the time duration for
switching from the
current VoIP link to
another VoIP link or a
PSTN link (that is, the
call backup switching
time) in case of a VoIP
call failure.
voip timer voip-to-pots time
By default, the duration of monitoring DSP
buffered data is 270 milliseconds. If the
duration is set to 0, buffered data will not be
monitored.
Duration greater than 240 milliseconds is
recommended because too small a
duration value will result in poor voice
quality in the case of severe jitter.
Optional.
The default is 101110.
Optional.
The default is 5 seconds.
The default is the general version.
6.
Configure the type of
the DSP image.
vi-card dsp-image { ms |
general }
7.
Configure the maximum
number of call history
records that can be
stored.
call-history max-count
number
The general version supports the G.723
codec, but it cannot meet voice quality
requirements of Microsoft. The ms version
can meet voice quality requirements of
Microsoft, but it does not support the G.723
codec.
Optional.
50 by default.
NOTE:
• In voice view, the vqa dscp media command has global significance, and the dscp media
command is valid only for the configured voice entity.
• The DSP image type setting takes effect after a reboot.
Configuring global default parameters for voice
entities
For each voice entity, if the certain command is not performed to configure a parameter, the system
uses the default value of the parameter.
When there are too many voice entities on one router and the values of most voice entity parameters
are the same as the default values, the default values can be adopted to improve the efficiency.
In cases where the manually configured parameter values of most voice entities are almost the same
but different from the default values, to specify suitable voice parameters for these voice entities one
by one will be time-consuming. In this case, you can use the default command to generate new
22
default values globally. In this way, each voice entity will directly inherit new default values, making
the configuration more flexible, simple and convenient.
Difference between the default command and the undo default command
To configure default values for parameters of voice entities globally, use the default command (such
as the default entity vad-on command).
To restore the global parameters to the defaults, use the undo default command.
Configuration procedure
To configure global default voice parameters:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enable VAD globally as the
default.
default entity vad-on
Optional.
Not enabled by default.
Optional.
5.
Specify the default global
codecs and their priority
levels.
default entity compression
{ 1st-level | 2nd-level | 3rd-level |
4th-level } { g711alaw |
g711ulaw | g723r53 | g723r63 |
g726r16 | g726r24 | g726r32 |
g726r40 | g729a | g729br8 |
g729r8 }
By default, the codec with the first
priority is g729r8, that with the
second priority is g711alaw, that
with the third priority is g711ulaw,
and that with the fourth priority is
g723r53.
The default entity compression
command takes no effect on IVR
voice entities.
Optional.
6.
Configure globally the default
voice packetization period for
different codecs.
default entity payload-size
{ g711 | g723 | g726r16 | g726r24
| g726r32 | g726r40 | g729 }
The default is 20 milliseconds for
a G.711 codec, and 30
milliseconds for G.723, G.726,
and G.729 codecs.
Enabling FXO monitoring
When FXO monitoring is enabled, the router displays the physical state as down for FXO ports that
are loosely connected or not connected at all.
In environments with instable PSTN lines or poor line quality, FXO monitoring might detect that some
lines are disconnected although those line are actually connected, resulting in call failures. To avoid
this problem, disable FXO monitoring.
To enable FXO monitoring:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enable FXO monitoring.
fxo-monitoring enable
Optional.
23
By default, FXO monitoring is
enabled.
Enabling the trap function
With the trap function enabled on a call module, when it originates a call, traps with the severity level
warning are generated by the module to inform the important events. These traps will be sent to the
information center of the device. You can set parameters for the information center to determine the
output rules of traps. For more information about the parameter settings of the information center,
see Network Management and Monitoring Configuration Guide.
You can enable the trap function either globally or for an entity. If it is enabled globally, traps will be
generated for all entities. If it is enabled only for a specified entity, traps are generated only for this
entity.
To enable the trap function globally:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enable the trap function
globally.
snmp-agent trap enable voice
dial
Optional.
Disabled by default.
To enable the trap function for an entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a POTS or VoIP voice
entity and enter its view.
entity entity-number { pots |
voip }
N/A
5.
Enable the trap function for
the entity.
dial-trap enable
Optional.
Disabled by default.
Displaying and maintaining voice entity
configuration
Task
Command
Remarks
Display voice call information.
display voice call-info { brief |
mark number | verbose } [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Display information about the call
management center (CMC)
module.
display voice cmc { ccb |
statistic [ all | em | h323 | iva |
lgs | r2 | sip | tmrout | vim ] } [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Display current default value and
system fixed default value.
display voice default all [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Display configuration information
about different types of voice
display voice entity { all | ivr |
mark entity-tag | pots | vofr |
Available in any view.
24
Task
Command
entities.
voip } [ | { begin | exclude |
include } regular-expression ]
Remarks
Display the jitter buffer statistics of
the last call.
display voice jitter-buffer
subscriber-line line-number [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Display the statistics of the active
calls.
display voice statistics
call-active { all | calling
calling-number | called
called-number } [ | { begin |
exclude | include }
regular-expression ]
Available in any view.
Display the call statistics of voice
entities after the system starts up.
display voice statistics entity
{ all | mark entity-index } [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Display the statistics of the IPP
module.
display voice ipp statistic { all |
cmc | h225 | h245 | ras | socket |
timer } [ | { begin | exclude |
include } regular-expression ]
Available in any view.
Display related information about
the IVA module.
display voice iva statistic { all |
call | cmc | error | isdn | proc |
timer | vim } [ | { begin | exclude |
include } regular-expression ]
Available in any view.
Display history records of the calls
that have ended.
display voice statistics
call-history { all | last index } [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Clear the call statistics of the CMC
module.
reset voice cmc statistic
Available in user view.
Clear the call statistics of the IVA
module.
reset voice iva statistic
Available in user view.
Clear the call statistics of the IPP
module.
reset voice ipp statistic
Available in user view.
Voice entity configuration examples
This section provides voice entity configuration examples.
Voice entity configuration example for establishing a VoIP
call
Network requirements
As shown in Figure 6, two voice gateways (Router A and Router B) communicate with each other
through WAN.
For example, the user of tel. 1 (010-1001) attached to Router A dials 0755-2001, the number of tel. 3
attached to Router B. After the called party picks up the phone, a conversation is established
between two parties.
25
Figure 6 Network diagram
Configuration procedure
Routing-related configurations are beyond the scope of this example. This example assumes that
Router A and Router B are reachable to each other.
1.
Configure Router A:
# Configure the VoIP voice entity to Router B.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
[RouterA-voice-dial-entity755] quit
# Configure the POTS voice entity corresponding to the local interface Line 1/0.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/0
[RouterA-voice-dial-entity1001] quit
# Configure the POTS voice entity corresponding to the local interface Line 1/1.
[RouterA-voice-dial] entity 1002 pots
[RouterA-voice-dial-entity1002] match-template 0101002
[RouterA-voice-dial-entity1002] line 1/1
2.
Configure Router B:
# Configure the VoIP voice entity to Router A.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Configure the POTS voice entity corresponding to the local interface Line 2/0.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity1001] match-template 07552001
[RouterB-voice-dial-entity1001] line 2/0
[RouterB-voice-dial-entity1001] quit
# Configure the POTS voice entity corresponding to the local interface Line 2/1.
26
[RouterB-voice-dial] entity 2002 pots
[RouterB-voice-dial-entity1002] match-template 07552002
[RouterB-voice-dial-entity1002] line 2/1
Fast connection
Network requirements
As shown in Figure 7, phone users of Router A and Router B communicate with each other over
WAN. The connection from Router A to Router B adopts fast connection and DTMF H.225
out-of-band transmission. The connection from Router B to Router A adopts DTMF out-of-band
transmission and fast connection.
Figure 7 Network diagram
Configuration procedure
Routing-related configurations are beyond the scope of this example. This example assumes that
Router A and Router B are reachable to each other.
1.
Configure Router A:
# Configure a VoIP voice entity.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address ip 2.2.2.2
# Enable fast connection, tunnel function and DTMF H.225 out-of-band transmission for the
VoIP voice entity.
[RouterA-voice-dial-entity755] fast-connect
[RouterA-voice-dial-entity755] tunnel-on
[RouterA-voice-dial-entity755] outband h245
[RouterA-voice-dial-entity755] quit
# Configure the local interface and phone number for Telephone A.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/1
# Enable out-of-band DTMF transmission for the POTS voice entity.
[RouterA-voice-dial-entity1001] outband h225
2.
Configure Router B:
# Configure the VoIP voice entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address ip 1.1.1.1
27
# Enable out-of-band DTMF transmission for the VoIP voice entity.
[RouterB-voice-dial-entity10] outband h225
[RouterB-voice-dial-entity10] quit
# Configure the local interface and phone number for Telephone B.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/1
# Enable out-of-band DTMF transmission for the POTS voice entity.
[RouterB-voice-dial-entity2001] outband h225
Troubleshooting voice entity configuration
Busy tone heard immediately after number dialed
Symptom
The busy tone is heard immediately after a number is dialed.
Solution
•
Check whether the called party is busy.
•
Check that the peer is reachable. You can use the ping command to ping the peer's IP address.
•
Check that the voice entity configuration is correct.
28
Configuring analog voice subscriber lines
This chapter covers the configuration of analog FXS, FXO, and E&M voice subscriber lines.
Signal tone
Call progress tones (CPTone), also called signal tones, are generally composed of several discrete
single-frequency tones that are played repeatedly on a make-break ratio basis. Signal tones,
including dial tone, ringback tone, and busy tone, are used to inform users of the call progress.
Signal tones vary greatly with countries as per their national standards.
Signal tones support two modes: country mode and custom mode. In addition, the amplitude of
signal tones can be customized to avoid signal tone detection failure.
FXS voice subscriber line
FXS interface
A foreign exchange station (FXS) interface uses a standard RJ-11 connector and a telephone cable
to directly connect to an ordinary telephone or a fax machine. An FXS interface accomplishes
signaling exchange based on the level changes on the Tip/Ring line and provides ring, voltage, and
dial tone.
CID
Calling identity delivery (CID) enables called terminals to display the calling identity information,
including the calling number, calling name, date, and time.
With the CID function, calling numbers and calling time in single-data-message format are
transmitted or received in an on-hook state. When the CID function is combined with services such
as call forwarding—unconditional (CFU) and call forwarding—busy (CFB), calling identity
information can also be transmitted if required. A message in the single-data-message format
contains the following information:
•
Date and time when the voice call occurs (MM DD hh:mm)
•
Calling number if CID is enabled on the device
•
P if CID is disabled on the device
•
O if the terminating PBX fails to obtain the calling number (for example, the originating PBX end
does not send it)
A message in the multiple-data-message format contains the following information:
•
Date and time when the voice call occurs (MM DD hh:mm)
•
Calling number and calling name if CID is enabled on the device
•
Two Ps for the calling number and the calling name if CID is disabled on the device
•
O if the terminating PBX fails to obtain the calling number (for example, the originating PBX end
does not send it)
•
O if the terminating PBX fails to obtain the calling name (for example, the originating PBX end
does not send it)
The FXS voice subscriber line sends the calling identity information to the called telephone. The
calling identity information is sent to the called telephone through frequency shift keying (FSK)
modulation between first and second rings. Therefore, the called user must pick up the telephone
29
after the second ring to make sure that the calling identity information is sent and received correctly.
Otherwise, the calling identity information might fail to be displayed.
FXO voice subscriber line
FXO interface
A foreign exchange office (FXO) interface, that is, a two-port loop trunk interface, uses an RJ-11
connector and a telephone cable to connect local calls to a PSTN or PBX. Like an FXS interface, an
FXO interface accomplishes signaling exchange based on the level changes on the Tip/Ring line. An
FXO interface can be connected only to an FXS interface.
CID
The FXO voice subscriber line receives the calling identity information from the PBX. The FXO
interface receives the modulation information of the calling identity information from the PBX
between the first and second rings. (This is the default situation. Use the cid ring command to
configure the time for CID check.) Then, the calling identity information undergoes FSK
demodulation and parity check. The function of sending calling identity information is checked after
the parity check succeeds. If the function is enabled, the calling identity information is sent to the IP
network. If the function is disabled, the identity information is empty.
Busy tone detection
Signaling standards and busy tone characteristics might vary with devices. Therefore, busy tones
cannot be recognized by a fixed threshold value.
The automatic busy tone recognition technology can solve this issue. This technology uses software
to sample and analyze busy tones to produce a set of parameters that represent the most common
patterns of busy tones. After configured with these parameters, the FXO ports can recognize
different types of busy tones.
Busy tone recognition includes the following elements:
•
Busy tone frequency—Most busy tones have one or two frequencies.
•
Duty ratio—Duration ratio of high/low levels composing a busy tone signal, which is also
referred to as make-break ratio. The specifications of the duty ratio of a busy tone vary with
countries and regions. The national standard of China is 350 milliseconds/350 milliseconds (10%
error allowed).
•
Detection threshold—Threshold used to determine whether a level is a high level or low level.
If the level higher than the threshold, it is regarded as a high level. Otherwise, it is regarded as
a low level.
Figure 8 shows the typical network diagram for automatic busy tone detection. Telephone A is
connected to PBX A, which connects to the FXO interface of Router A through an ordinary telephone
line. The connection of the peer end is similar.
30
Figure 8 Network diagram
Perform an automatic busy tone detection test as follows:
1.
Dial number 1002 from Telephone A (010-1001). The FXO interface on Router A plays a dial
tone to PBX A, which then transmits the tone to Telephone A. Then dial number 07552001 from
Telephone A. Telephone B rings. After Telephone B is picked up, the call is connected.
2.
If you hang up Telephone A, PBX B plays a busy tone to Router A.
3.
Use the vi-card busy-tone-detect command in voice view to start the detection. To ensure the
detection of the busy tone signal sent by PBX B, H3C recommends that you run the command
2 seconds after on-hook.
4.
(The console terminal displays "Begin to autodetect busy-tone, it will take about 12 seconds,
please wait...", which indicates that the busy tone detection is in progress. After the detection,
the terminal displays "Auto-detect busy-tone succeeded!", which indicates that the detection
succeeded.) Use the save command to save the detected busy tone parameters.
5.
Repeat step 2 to verify whether the automatic busy tone detection succeeded. If so, Telephone
A will automatically be disconnected after PBX B plays busy tones to Router A for 4 seconds.
E&M subscriber line
E&M interface
An ear & mouth or receive & transmit (E&M) interface uses a RJ-48 telephone cable to connect a
PBX. The PBX sends signals on the M (mouth) line and receives signals on the E (ear) line. The
voice router receives M signals from the PBX and sends E signals to the PBX. An E&M interface can
only be connected to another E&M interface.
When E&M is applied in VoIP communication, two or four voice wires can be used. There are two or
four signaling wires as well. Therefore, 4-wire analog E&M actually has six wires. In two-wire mode,
voice is transmitted over each wire in each direction simultaneously. In four wire mode, voice is
transmitted over each wire pair in each direction simultaneously.
E&M start mode
An E&M interface supports E&M signaling and divides each voice connection into trunk circuit side
and signaling unit side (similar to DCE and DTE).
An E&M interface provides on-hook/off-hook signals with minimum interference. Instead of using dial
tones, E&M interfaces use one of the following main start dial supervision signaling protocols:
•
Immediate start—As shown in Figure 9, the calling side goes off-hook, waits for a certain
period of time, and sends the dialed digits regardless of whether the called side is ready or not.
If the called side receives the digits, it rings to alert the called party of the call. If the called party
picks up the phone, the call is connected.
31
Figure 9 Immediate start mode
•
Delay start—As shown in Figure 10, the calling side goes off-hook to seize the trunk and the
called side (PBX) goes off-hook to respond to the seizure. When the called side (PBX) is ready,
it goes on-hook. The off-hook interval is the delay dial signal. Then, the calling side starts
sending the address information and the PBX at the called side routes the call to its destination.
When the called party answers, the called side goes off-hook and both sides remain off-hook for
the duration of the call.
Figure 10 Delay start mode
•
Wink start—As shown in Figure 11, the calling side goes off-hook to seize the trunk. The called
side (PBX) remains on-hook until it receives the connection signal from the calling side. Once
the called side is ready, it sends a wink signal. Once the calling side receives the wink, it sends
the address information, and the call is routed to its destination. When the called party answers,
the called side goes off-hook and both sides remain off-hood for the duration of the call.
Figure 11 Wink start mode
Hardware compatibility with analog voice
subscriber lines
Analog voice subscriber lines are not available on the following routers:
32
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
Configuration task list
Task
Remarks
Configuring call progress tones
Required.
Configuring basic functions
Optional.
Configuring FXS voice subscriber line
Optional.
Configuring FXO voice subscriber line
Optional.
Binding an FXS voice subscriber line to an FXO voice subscriber line
Optional.
Configuring E&M voice subscriber line
Optional.
Configuring DTMF
Optional.
Configuring options related to dial plan
Optional.
Configuring adjustment functions
Optional.
Rebooting a voice card
Optional.
Configuring global default parameters for voice subscriber lines
Optional.
Configuring call progress tones
This section covers how to specify and configure the call progress tones of a country.
Configuration prerequisites
The router is equipped with an applicable FXS, FXO, E&M, BSV, VE1, or VT1 voice interface card.
Specifying the call progress tones of a country
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Specify the call progress
tones of a country.
cptone country-type locale
By default, the call progress tones
of China are specified.
4.
Configure the level
parameter for the call
cptone tone-type { all |
Optional.
33
Step
progress tones.
Command
Remarks
busy-tone | congestion-tone |
dial-tone | ringback-tone |
special-dial-tone |
waiting-tone } amplitude value
By default, 1000 for busy tone and
congestion tone, 400 for dial tone
and special dial tone, and 600 for
ringback tone and waiting tone.
NOTE:
The configuration of the cptone country-type command will take effect on all voice ports of all cards
on the device.
Customizing call progress tones for a country
You can customize the call progress tones of a country if necessary.
To customize call progress tones for a country:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Customize call
progress tone
parameters.
vi-card cptone-custom { busy-tone |
congestion-tone | dial-tone |
ringback-tone | special-dial-tone |
waiting-tone } comb freq1 freq2 time1
time2 time3 time4
By default, no call progress tones
are customized.
4.
Select a country.
cptone country-type CS
By default, the country mode is
China.
5.
Configure the level
parameter for the
customized call
progress tones.
cptone tone-type { all | busy-tone |
congestion-tone | dial-tone |
ringback-tone | special-dial-tone |
waiting-tone } amplitude value
Optional.
By default, 1000 for busy tone and
congestion tone, 400 for dial tone
and special dial tone, and 600 for
ringback tone and waiting tone.
Configuring basic functions
This section covers prerequisites and procedures for configuring basic functions.
Configuration prerequisites
The router is equipped with an applicable FXS, FXO, E&M, or BSV voice interface card.
NOTE:
The commands in FXS, FXO, E&M, or BSV voice subscriber line view are applicable only when the
router is equipped with an FXS, FXO, or E&M interface card.
Configuration procedure
This section describes the basic functions of voice subscriber lines generated by voice cards.
Configurations specific to a voice subscriber line will be introduced individually.
34
To complete the basic functions of voice subscriber lines:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Set the expected bandwidth
for the voice subscriber line.
bandwidth bandwidth-value
4.
Configure a voice
subscriber-line description.
description string
By default, the description string
of a voice subscriber line is
interface-name+interface.
5.
Restore the default
configuration of the voice
subscriber line.
default
Optional.
Tear down the voice
subscriber line.
shutdown
Optional.
Support for the command
depends on your device model.
Optional.
6.
Optional.
Up by default.
Configuring FXS voice subscriber line
This section covers prerequisites and procedures for configuring an FXS voice subscriber lines.
Configuration prerequisites
•
The router is equipped with an applicable FXS interface card.
•
The basic functions of FXS voice subscriber lines are configured.
Configuration guidelines
•
To implement the CID function, the VoIP router must be capable to receive calling identity
information in the on-hook state, the hardware and software of PBX must support this service,
and the called telephone must be able to receive and display calling identity information (such
telephones are CID I or CID II class) in the on-hook or conversation state.
•
The call date and time transmitted in data-message format is the router system time. To ensure
correct call time, use the clock command to synchronize router time with the local standard
time.
Configuring CID
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXS voice
subscriber line view.
subscriber-line line-number
N/A
3.
Enable CID on the FXS
voice subscriber line.
cid display
Enable the FXS voice
cid send
4.
Optional.
Enabled by default.
Optional.
35
Step
Command
Remarks
subscriber line to send
calling identity
information to the remote
end.
5.
6.
7.
Configure the calling
name for the FXS voice
subscriber line.
Enabled by default.
calling-name text
Optional.
Not configured by default.
Optional.
Configure the format of
messages (which carry
the calling number
information) transmitted
over the FXS voice
subscriber line.
cid type { complex | simple }
When the peer end supports only one
message format, you must use the
same message format at the local end
as the one used at the peer end.
Configure the standard
for the FXS voice
subscriber line to send
messages carrying the
calling number
information.
cid standard-type { bellcore |
brazil }
Optional.
The default is complex.
By default, bellcore is adopted.
Configuring packet loss compensation mode
Step
Command
Remarks
1.
Enter system view.
System-view
N/A
2.
Enter FXS voice subscriber
line view.
subscriber-line line-number
N/A
3.
Configure the packet loss
compensation mode for the
FXS voice subscriber line.
Optional.
plc-mode { general | specific }
By default, the specific algorithm
provided by the voice gateway is
configured for the FXS voice
subscriber line.
Setting the electrical impedance
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXS voice subscriber
line view.
subscriber-line line-number
N/A
Set the electrical impedance.
impedance { country-name | r550
| r600 | r650 | r700 | r750 | r800 |
r850 | r900 | r950 }
3.
Optional.
By default, the electrical
impedance is the impedance
value applicable to China.
Configuring the sending of LCFO signals
You can configure an FXS voice subscriber line to send a loop current feed open (LCFO) signal to
indicate a disconnection to the peer. This feature is used mainly in North America.
To configure the sending of LCFO signals:
36
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXS voice subscriber
line view.
subscriber-line line-number
N/A
3.
Enable the FXS voice
subscriber line to send an
LCFO signal at hangup.
disconnect lcfo
Configure the LCFO signal
duration.
timer disconnect-pulse
milliseconds
Optional.
4.
By default, an FXS voice
subscriber line does not send
an LCFO signal at hangup. It
plays busy tones to the peer.
Optional.
750 milliseconds by default.
Configuring FXO voice subscriber line
This section covers the prerequisites and procedures for configuring an FXO voice subscriber line.
Configuration prerequisites
•
The router is equipped with an applicable FXO interface card.
•
The basic functions of FXO voice subscriber lines are configured.
Configuration guidelines
•
To implement the CID function, the VoIP router must be capable to receive calling identity
information in the on-hook state, the hardware and software of PBX must support this service,
and the called telephone must be able to receive and display calling identity information (such
telephones are CID I or CID II class) in the on-hook or conversation state.
•
The call date and time transmitted in data-message format is the router system time. To ensure
correct call time, use the clock command to synchronize router time with the local standard
time.
Configuration procedure
To enable calling identity information receiving and sending:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice subscriber line view.
subscriber-line
line-number
N/A
3.
Enable calling identity information
receiving for the FXO voice
subscriber line.
cid receive
Enable calling identity information
sending for the FXO voice subscriber
line.
cid send
Configure the time for CID check and
after the CID check, the number of
rings the FXO line receives before
cid ring { 0 | 1 | 2 } [ times ]
4.
5.
Optional.
Enabled by default.
Optional.
Enabled by default.
Optional.
37
By default, CID check is
performed between the first
Step
Command
going off-hook.
Remarks
and the second rings and the
FXO line goes off-hook as
soon as the check completes,
that is, cid ring 1 0.
Configuring busy tone detection
This section describes how to configure busy tone detection for FXO voice subscriber lines.
Configuring busy tone detection parameters for the FXO voice subscriber line
The actual specifications of busy tones might differ from the configured parameters. If this is the case,
busy tones cannot be recognized, resulting in failed or false on-hook.
To configure the busy tone detection parameters for the FXO voice subscriber line:
Step
Command
Remarks
1.
Enter system view.
System-view
N/A
2.
Enter voice view.
voice-setup
N/A
Configure busy tone
detection parameters.
vi-card busy-tone-detect { auto
index line-number | custom
area-number index argu f1 f2 p1
p2 p3 p4 p5 p6 p7 }
Optional.
3.
The system can record four busy
tones which are identified by the
index argument.
Optional.
4.
Configure the type of busy
tone.
area { custom | europe |
north-america }
By default, the busy tone
compliant with the European
standard is used.
Once you execute this command,
the configuration will be applied to
all FXO voice cards on the device.
5.
Quit voice view.
quit
N/A
6.
Enter FXO voice subscriber
line view.
subscriber-line line-number
N/A
Optional.
The default is 2.
7.
Configure the number of
busy tone periods for
detection.
busytone-t-th time-threshold
You can increase the number of
busy tone detection periods by
using the busytone-t-th
command to improve the busy
tone detection accuracy. This
reduces the likelihood of false
on-hook, but increases the
likelihood of failed on-hook.
NOTE:
Before you configure the number of busy tone detection periods, test the new value repeatedly to
make sure that the new value does not cause failed or false on-hook.
Enabling the busy tone sending
If the PBX fails to play a busy tone to a digital telephone, enable the FXO interface to send a busy
tone to the PBX, which will transparently send the busy tone to the digital telephone.
38
To enable the busy tone sending:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice subscriber
line view.
subscriber-line line-number
N/A
Optional.
3.
Enable the busy tone
sending.
send-busytone { enable | time
seconds }
Disabled by default.
The time option appears only
after the send-busytone enable
command is executed.
Enabling silence detection-based automatic on-hook
Silence detection-based automatic on-hook prevents the situations where the resource of the FXO
interface cannot be released owing to busy tone detection failure when the busy tone parameters
provided by the connected PBX are special.
When the signal values of two successive sampling points are less than the silence threshold, the
system considers that the line goes into the silent state. If the line stays in the silent state longer than
the silence duration for automatic on-hook, the system will automatically disconnect the call.
Do not enable silence detection-based automatic on-hook unless you are sure it is necessary.
Improper configuration of this function can lead to false on-hook.
To enable silence detection-based automatic on-hook:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice subscriber
line view.
subscriber-line line-number
N/A
3.
Enable silence
detection-based automatic
on-hook.
Optional.
By default, the silence threshold is
3 and the silence duration for
automatic on-hook is 7200
seconds (2 hours).
silence-th-span threshold
time-length
NOTE:
It is a good practice to test multiple sets of parameters and choose the set of parameters that can
quickly release the FXO voice subscriber line after on-hook and does not cause false on-hook.
Configuring the duration before a forced on-hook
In some countries, PBXs do not play busy tones, or the busy tones played only last for a short period
of time. When noise is present on a transmission link, the silence-th-span command might fail to
release the FXO interface after on-hook. In this case, use the hookoff-time command to solve the
problem.
To configure the duration before a forced on-hook:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice
subscriber line view.
subscriber-line line-number
N/A
3.
Configure the duration
hookoff-time time
Optional.
39
Step
Command
Remarks
before a forced on-hook.
Forced on-hook is disabled by default.
The configuration will take effect on all
interfaces of a card after this command
is executed.
IMPORTANT:
Once forced on-hook is enabled, calls will be automatically disconnected when the duration expires,
even if the conversation is going on.
Configuring the delay time before an on-hook
Usually, after the FXO interface detects a busy tone, the system automatically disconnects the call
and immediately removes the connection. When an FXO subscriber line is used as the VoIP access
port can cooperate with an IP phone, because the IP phone does not play any prompt tone to the IP
phone user, it is easily for the IP phone user to ignore the busy tone and considers that the line failure
occurs when the FXO subscriber line detects the busy tone and removes the connection quickly.
With the delay time before an on-hook configured, when the FXO subscriber line detects a busy tone,
it waits for a period of time, and then disconnects a call and removes the connection. In this case, the
busy tone is first sent to the FXO interface and then sent to the IP phone, and the IP phone user will
easily confirm the busy tone information before the connection is removed.
To configure the delay time before an on-hook for the FXO voice subscriber line:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice
subscriber line view.
subscriber-line line-number
N/A
3.
Configure the delay time
before an on-hook for the
FXO voice subscriber
line.
busytone-hookon timer
seconds
Optional.
The default delay time is 0 seconds.
Configuring the off-hook mode
There are two off-hook modes after the FXO voice subscriber line receives ringing:
•
Immediate mode—The calling party dials the number of the trunk. Upon receiving the call, the
FXO goes off-hook and sends a dial tone to the calling party. Then, the calling party dials the
destination number.
•
Delay mode—The number of the private line is configured in subscriber line view on the
system. When the calling party dials the destination number, the call is routed to its destination
based on the configured private line number. When the called party picks up the phone, the
FXO goes off-hook for the conversation.
To configure the off-hook mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice subscriber
line.
subscriber-line line-number
N/A
3.
Configure the off-hook mode.
hookoff-mode { delay |
immediate }
Optional.
40
Immediate mode by default.
Setting ring detection parameters
PBXs from different vendors might use different types of ring signals. By setting ring detection
parameters, you can detect ring signals of different frequencies and waveforms.
To configure ring detection parameters:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice subscriber
line.
subscriber-line line-number
N/A
Optional.
3.
4.
Set the debounce time of ring
detection on the FXO voice
subscriber line.
ring-detect debounce value
Set the frequency value in
the ring detection.
ring-detect frequency value
10 milliseconds by default.
•
Do not set the debounce time
during a conversation.
•
Do not set the debounce time
too short or false ring tone
recognition might occur
under power line
interference.
•
If you configure this
command on a FXO voice
subscriber line of a board,
the configuration is effective
on all FXO subscriber lines
on this board.
Optional.
40 Hz by default.
Configuring other functions
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice subscriber
line view.
subscriber-line line-number
N/A
3.
Set the electrical impedance.
impedance { country-name | r550
| r600 | r650 | r700 | r750 | r800 |
r850 | r900 | r950 }
4.
Configure the packet loss
compensation mode.
plc-mode { general | specific }
Optional.
By default, the electrical
impedance conforms to Chinese
standards.
Optional.
The default is specific.
Binding an FXS voice subscriber line to an FXO
voice subscriber line
The one-to-one binding between FXS voice subscriber lines and FXO voice subscriber lines
enhances the reliability of voice communication, which is particularly important for some industries.
After the binding, FXO voice subscriber lines can be used for communication over PSTN when the IP
network is unavailable.
41
The one-to-one binding between FXS voice subscriber lines and FXO voice subscriber lines
provides the following functions:
•
Dedicated FXO voice subscriber lines—The FXO voice subscriber lines are used only for the
bound FXS voice subscriber lines and PSTN-originated calls received over the FXO voice
subscriber lines are directly connected to the bound FXS voice subscriber lines.
•
Consistent state between bound FXS and FXO voice subscriber lines—The
on-hook/off-hook state of the bound FXS and FXO voice subscriber lines is consistent. If an
FXS voice subscriber line receives a PSTN-originated call when the corresponding FXS voice
subscriber line goes off-hook, the calling party will hear busy tones.
Configuration prerequisites
The router is equipped with FXO and FXS interface cards. The basic functions of FXO and FXS
voice subscriber lines are configured.
Configuration procedure
To bind one FXS voice subscriber line to one FXO voice subscriber line:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO voice
subscriber line view.
subscriber-line line-number
N/A
3.
Bind an FXS voice
subscriber line to the FXO
voice subscriber line.
hookoff-mode delay bind
fxs_subscriber_line
[ ring-immediately ]
By default, no FXS voice
subscriber line is bound.
4.
Enable the automatic
dialing of the bound FXS
voice subscriber line.
private-line string
Disabled by default.
Configure an interval
between on-hook and
off-hook.
timer hookoff-interval milliseconds
6.
Exit FXO voice subscriber
line view.
quit
N/A
7.
Enter voice view.
voice-setup
N/A
8.
Enter voice dial program
view.
dial-program
N/A
9.
Create a POTS entity and
enter POTS entity view.
entity entity-number pots
N/A
5.
10. Specify the FXO voice
subscriber line to be
exclusively used by the
bound FXS voice
subscriber line.
Optional.
The default is 500 milliseconds.
Optional.
caller-permit calling-string
By default, no calling number is
configured. That is, incoming
calls are not restricted.
Configuring E&M voice subscriber line
Configuration prerequisites
•
The router is equipped with an applicable E&M interface card.
42
•
The basic functions of E&M voice subscriber lines are configured.
Configuring cable type
The configuration of the cable between a voice router and its peer device affects only voice
transmission and has no impact on signaling. Improper configuration can reduce the two-way
communication to a one-way voice communication.
To configure the cable type for the analog E&M trunk:
Step
Command
Remarks…
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure the cable type for
the E&M voice subscriber
line.
em-phy-parm { 2-wire | 4-wire }
The default is 4-wire.
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure a signal type for
the E&M voice subscriber
line.
type { 1 | 2 | 3 | 5 }
The default is 5 (corresponding V
type signal).
Configuring signal type
Configuring start mode
To configure the immediate start mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure the immediate
start mode for the E&M voice
subscriber line.
em-signal immediate
The default is immediate start
mode by default.
Configure a delay before the
originating side sends DTMF
signals in the immediate start
mode.
delay send-dtmf milliseconds
4.
Optional.
The default is 300 milliseconds.
To configure the delay start mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line view.
subscriber-line
N/A
43
Step
Command
Remarks
line-number
3.
Configure the delay start mode for the
E&M voice subscriber line.
em-signal delay
4.
Configure the delay signal duration in
the delay start mode.
delay hold milliseconds
5.
Configure the delay time from when
the terminating side detects a seizure
signal to when it sends a delay signal
in the delay start mode.
delay rising milliseconds
N/A
Optional.
The default is 400
milliseconds.
Optional.
The default is 300
milliseconds.
To configure the wink start mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line view.
subscriber-line line-number
N/A
3.
Configure the wink start mode for
the E&M voice subscriber line.
em-signal wink
The default is immediate start
mode.
4.
Configure the delay time from
when the terminating side
receives a seizure signal to when
it sends a wink signal in the wink
start mode.
delay send-wink milliseconds
Configure the maximum amount
of time the originating side waits
for a wink signal after sending a
seizure signal in the wink start
mode.
delay wink-rising milliseconds
Configure the duration of a wink
signal sent by the terminating side
in the wink start mode.
delay wink-hold milliseconds
5.
6.
Optional.
The default is 200 milliseconds.
Optional.
The default is 2,000
milliseconds.
Optional.
The default is 500 milliseconds.
Enabling E&M non-signaling mode
The E&M non-signaling mode is applied when the E&M interface of the peer device does not provide
the M line and E line. In this mode, the E&M interface communicates with the peer end without
signaling. You can configure the private line auto ring-down (PLAR) function by using the
private-line command to form a three-segment E&M virtual private line (E&M-VoIP-E&M). When a
subscriber picks up the phone, a number is automatically dialed out through the E&M virtual private
line.
Configuration guidelines
•
This feature should be used with the PLAR function, and you must execute the private-line
command on the calling voice gateway. For more information about the PLAR function, see
"Configuring dial plans."
•
Before you enable E&M non-signaling mode, the E&M signaling must operates in the
immediate start mode.
•
For more information about digital E&M non-signaling mode, see "Configuring digital voice
subscriber lines."
44
Configuration procedure
To enable E&M non-signaling mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line view.
subscriber-line line-number
N/A
3.
Enable E&M non-signaling mode.
open-trunk { caller monitor
interval | called }
Optional.
Disabled by default.
Enabling E&M analog control signals pass-through
This feature operates only when E&M has no signaling enabled. As shown in Figure 12, the Tone
Generator and the radio establish an E&M virtual private line. Enable E&M analog control signals
pass through for Router A and Router B so that they can send occupied and idle signals for the E&M
virtual line over the IP network.
Figure 12 E&M analog control signals pass-through
To enable E&M analog control signals pass-through:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable E&M analog control
signals pass-through.
em-passthrough
Optional.
Disabled by default.
NOTE:
Configure this feature on the voice gateways of both sides.
Configuring analog E&M line failure tone
As shown in Figure 12, E&M analog control signals pass-through is enabled for the analog E&M line
on Router A. To notify an IP network failure or a peer failure to the Tone Generator, configure Router
A to play busy tones for analog E&M line failure by using the em-failure busytone command.
Before performing this task, make sure that both the E&M non-signaling mode and the E&M analog
control signals pass-through are enabled for the E&M line. For more information, see "Enabling E&M
non-signaling mode" and "Enabling E&M analog control signals pass-through."
To configure analog E&M line failure tone:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
45
Step
3.
Command
Configure analog E&M line
failure tone.
Remarks
em-failure { busytone | silence }
Optional.
The default setting is silence.
Configuring output gain of SLIC chip
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure the output gain of
the SLIC chip.
slic-gain { 0 | 1 }
Optional.
The default is 0 (0.8 dB).
Configuring DTMF
Introduction to DTMF
Dual tone multi-frequency (DTMF) uses a mixture of a high frequency tone and a lower frequency
tone to represent a key on a keypad. Each column of keys is represented by a high frequency tone
and each row of keys is represented by a low frequency tone. For example, as shown in Figure 13,
the digit 1 is represented by the combination of a pure 697 Hz signal and a pure 1209 Hz signal.
Such DTMF signals have good immunity to interference.
Figure 13 DTMF keypad frequencies
Column Frequency Group
1209Hz
1336Hz
1477Hz
1633Hz
697Hz
1
2
3
A
770Hz
4
5
6
B
852Hz
7
8
9
C
941Hz
*
0
#
D
A DTMF signal must last at least 45 milliseconds. A minimum interval of 23 milliseconds is required
between two DTMF signals to make sure that DTMF signals are recognizable. Such requirements
46
are roughly the same in all countries. For more information, refer to the ITU-T Recommendation
Q.24.
Configuring DTMF properties
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure the duration of
DTMF tones and the interval
between successive DTMF
tones.
dtmf time {interval | persist }
milliseconds
Configure a DTMF signal
amplitude.
dtmf amplitude value
4.
Optional.
By default, 120 milliseconds for
both.
Optional.
The default is –9.0 dBm.
NOTE:
The dtmf time and dtmf amplitude commands in voice view have global significance. Once you
carry out either of the two commands, the configuration will take effect on the whole device.
Configuring DTMF detection
Introduction to DTMF
Use the following ways to detect DTMF:
•
Energy detection—DTMF detection is implemented by calculating the frequency spectrum of
the input voice signal. The energy threshold limits the spectrum shape of the input signal. A
signal is considered a valid DTMF only when all requirements are met.
•
Sensitivity detection—A higher DTMF detection sensitivity reduces the possibility of missing
the detection of a true DTMF signal but increases the possibility of false detection; a lower
DTMF detection sensitivity reduces the possibility of false detection but increases the possibility
of missing the detection of a true DTMF signal.
Configuration procedure
To configure DTMF detection:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber
line view.
subscriber-line line-number
N/A
Optional.
3.
Configure the DTMF
detection sensitivity.
dtmf threshold analog index
value
By default, indexes 0 to 12 correspond
to 1400, 458, -9, -9, -9, -9, -3, -12, -12,
30, 300, 3200, and 375, respectively.
For meanings of these parameters, see
Voice Command Reference.
This command is used by professional
personnel to adjust the device in the
case of DTMF detection failure. Usually,
the default value is adopted.
4.
Configure the DTMF
detection sensitivity
dtmf sensitivity-level { high |
low | medium
47
Optional.
Step
level.
Command
Remarks
[ frequency-tolerance value ] }
By default, the DTMF detection
sensitivity is low.
This command is valid only for the
FXS/FXO voice subscriber line.
Support for the medium keyword
depends on the voice card.
Optional.
5.
Set the DTMF detection
sensitivity.
dtmf threshold digital value
By default, the value of DTMF detection
sensitivity level is 0, that is, insensitive.
This command is only applicable to the
BSV voice subscriber line.
Configuring options related to dial plan
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable the private line auto
ring function.
private-line string
Bind a calling/called number
substitution rule list to a voice
subscriber line.
substitute { called | calling }
list-number
4.
Optional.
Disabled by default.
By default, no number substitution
rule list is bound to a voice
subscriber line.
For more information about the operation of the above commands, see "Configuring dial plans."
Configuring adjustment functions
Configuration task list
Task
Remarks
Configuring echo cancellation
Optional.
Configuring gain adjustment function
Optional.
Configuring time adjustment function
Optional.
Configuring comfortable noise function
Optional.
Configuring PCM pass-through function
Optional.
Configuring echo cancellation
Echo cancellation
An echo is the audible leak-through of your own voice into your own receive path. When the voice of
a user leaks into the receive path of the user, it is an echo. To cancel echoes, adjust the echo
cancellation function in the VoIP gateway as follows:
48
1.
Adjust echo duration
Table 2 Adjust echo duration
Symptom
A user hears a delayed
copy of the original
voice of the user.
2.
Reason
Adjustment method
The echo duration is so long that the
convergence time of echo cancellation on the
network becomes longer.
Shorten echo duration
The echo duration is so short that long-duration
echoes are not completely cancelled.
Prolong echo duration
Adjust echo cancellation parameters
Table 3 Adjust echo cancellation parameters
Symptom
Parameters adjusted
Effect
A user hears echoes or loud
background noises from the peer
when speaking.
Speed up the convergence of
comfortable noise amplitudes
Too fast convergence might make
noises uncomfortable.
There are loud environment
noises.
Increase the maximum amplitude
of comfortable noises.
Too large amplitude might make
noises uncomfortable.
A user hears echoes when
speaking.
Enlarge the control factor of mixed
proportion of noises.
Too high a control factor leads to
audio discontinuity.
There are echoes when both
parties speak at the same time.
Enlarge the judgment threshold
for bidirectional conversation.
Too high a judgment threshold
slows down the convergence of
the filter factor.
3.
Enable the nonlinear function of echo cancellation
The nonlinear function of echo cancellation, also known as residual echo suppression, means
the removal of residual echoes after echo cancellation when the user at the local end does not
speak.
Configuration procedure
To configure the echo adjustment function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
Optional.
By default, the convergence rate
of comfort noise amplitude is 0,
the maximum amplitude of
comfort noise is 256, the comfort
noise mixture proportion control
factor is 100, and the threshold of
two-way talk is 1.
3.
Configure echo cancellation
parameters.
echo-canceller parameter
{ convergence-rate value |
max-amplitude value |
mix-proportion-ratio value |
talk-threshold value }
4.
Exit voice view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
6.
Enable the echo cancellation
function.
echo-canceller enable
Enabled by default.
7.
Configure echo duration.
echo-canceller tail-length
milliseconds
Optional.
49
The default is 0 milliseconds.
Step
Command
Remarks
Optional.
8.
Enable the nonlinear function
of echo cancellation.
Enabled by default.
nlp-on
This command is available only
after the echo-canceller enable
command is executed.
Configuring gain adjustment function
Voice subscriber line
To configure the gain adjustment function on a voice subscriber line:
Step
Command
Remarks…
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Set the input gain on the
voice interface.
receive gain value
Set the output gain on the
voice interface.
transmit gain value
4.
Optional.
The default is 0 dB.
Optional.
The default is 0 dB.
IMPORTANT:
Gain adjustment might lead to call failures. H3C recommends not adjusting the gain. If necessary,
make sure you understand the impact of the adjustment on your call before you adjusting the gain.
MoH/Paging voice subscriber line
To configure the gain adjustment function on the music on hold (MoH) or paging voice subscriber
line:
Step
Command
Remarks…
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Set the input gain on the
voice interface.
Optional.
audio-input-gain value
The default is 27.5 dB.
This command is only applicable
to the MoH voice subscriber line.
Optional.
4.
Set the output gain on the
voice interface.
The default is -10.0 dB.
audio-output-gain value
This command is only applicable
to the paging voice subscriber
line.
Optional.
5.
Mute the audio interface.
audio-mute
By default, an audio interface is
not muted.
This command is only applicable
to the MoH/paging voice
50
Step
Command
Remarks…
subscriber line.
IMPORTANT:
• Setting gains might decrease the quality of the input/output audio. H3C recommends not
adjusting the gain. If necessary, make sure you understand the impact of the adjustment before
you adjusting the gain.
• An audio interface that is muted with the audio-mute command cannot transmit data.
Configuring time adjustment function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure the interval
between off-hook and dialing
the first digit.
timer first-dial seconds
Configure the maximum
interval for dialing the next
digit.
timer dial-interval seconds
Configure the maximum
duration of playing ringback
tones.
timer ring-back seconds
4.
5.
Optional.
The default is 10 seconds.
Applicable only in FXO/FXS voice
subscriber line view.
Optional.
The default is 10 seconds.
Optional.
The default is 60 seconds.
Optional.
6.
7.
Configure a dial delay.
Configure the maximum
duration the system waits for
a digit.
delay start-dial seconds
The default is 1 second.
Applicable only in FXO/FXS voice
subscriber line view.
Optional.
timer wait-digit { seconds |
infinity }
The default is 5 seconds.
Applicable only in E&M voice
subscriber line view.
Optional.
8.
Configure the time range for
the duration of an on-hook
condition that will be
detected as a hookflash.
timer hookflash-detect
hookflash-range
By default, the time range is 50 to
180 milliseconds, that is, if an
on-hook condition that lasts for a
period that falls within the
hookflash duration range is
considered a hookflash.
Applicable only in analog FXS
voice subscriber line view.
Configuring comfortable noise function
Use the cng-on command to generate some comfortable background noise to replace the toneless
intervals during a conversation. If no comfortable noise is generated, the toneless intervals will make
both parties in conversation feel uncomfortable.
51
To configure the comfortable noise function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable the comfortable noise
function.
cng-on
Optional.
Enabled by default.
Configuring PCM pass-through function
To reduce internal processing delay, configure the PCM pass-through function.
Configuration restrictions and guidelines
•
This function is supported on E&M cards only. You must specify an E&M card when you
configure this function.
•
This function takes effect only when G.711 A-Law is used.
•
You must reboot the specified card after you configure this function. To check whether this
function takes effect, use the display device verbose command.
Configuration procedure
To configure the PCM pass-through function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enable the PCM
pass-through function.
pcm-passthrough slot
slot-number
Optional.
By default, the PCM pass-through
function is disabled.
Rebooting a voice card
Rebooting a voice card initializes the card and interrupts all traffic on it. You can reboot a voice card
when it fails (for example, the configuration is correct but no call connection can be established). The
reboot process is indicated by the LED.
Configuration guidelines
•
The vi-card reboot command reboot analog voice cards (including FXS, FXO, and E&M),
SIC-AUDIO, and BSV.
•
The SIC digital voice cards and VE1 and VT1 voice cards cannot be rebooted by using
commands.
•
FIC analog voice cards can also be rebooted by using the reboot slot slot-number command.
For more information about the reboot slot command, see Fundamentals Command
Reference.
52
Configuration procedure
To reboot a card:
Step
Command
Remarks
1.
Reset the voice card in user
view.
reboot slot slot-number
Optional.
2.
Enter system view.
system-view
N/A
3.
Enter voice view.
voice-setup
N/A
4.
Reboot a voice card.
vi-card reboot slot-number
Optional.
Configuring global default parameters for voice
subscriber lines
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure the default
input/output gain for all
subscriber lines.
default subscriber-line { receive
| transmit } gain value
Optional.
The default is 0.
For how to use the global default parameters for voice subscriber lines, see "Configuring voice
entities."
Mirroring PCM, RTP, or voice command data on
an analog voice subscriber line
The mirroring function copies the specified pulse code modulation (PCM), voice Real Time Protocol
(RTP), and voice command data on an analog voice subscriber line to the specified destination. With
the mirroring function, you can analyze and locate problems.
To mirror PCM, RTP, or voice command data on an analog voice subscriber line to a specified
interface or destination:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter analogy voice
subscriber line view.
subscriber-line line-number
N/A
3.
Mirror PCM, RTP packets, or
voice command data to a
specified interface or
destination.
mirror number number { pcm |
{ in | out | all } { command |
data } } to { local-interface
interface-type interface-number
[ mac H-H-H ] | remote-ip
ip-address [ port port ] }
53
Optional.
By default, no traffic is mirrored.
Displaying and maintaining analog voice
subscriber lines
Task
Command
Remarks
Display analog voice
subscriber-line information.
display voice subscriber-line
[ line-number ] [ brief ] [ | { begin |
exclude | include }
regular-expression ]
Available in any view.
Analog voice subscriber line configuration
examples
This section contains examples on configuring the FXO voice subscriber line and one to one binding
between FXS and FXO lines.
Configuration example for the FXO voice subscriber line
Network requirements
In the following figure, the FXO voice subscriber line connected to Router B operates in the
private-line auto ring-down (PLAR) mode, and the default remote phone number is 010-1001.
Dialing the number 0755-2003 on phone 0755-2001 connects to Router B. Since Router B operates
in the private-line mode, it requests connection to the preset remote number 010-1001 at Router A.
Figure 14 Network diagram
Configuration procedure
Routing-related configurations are beyond the scope of this example. This example assumes that
Router A and Router B are reachable to each other.
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
[RouterA-voice-dial-entity755] quit
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
54
[RouterA-voice-dial-entity1001] line 1/0
2.
Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/0
[RouterB-voice-dial-entity2001] send-number all
# Configure FXO interface Line 1/0.
[RouterB-voice-dial-entity2001] quit
[RouterB-voice-dial] quit
[RouterB-voice] quit
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] private-line 0101001
Configuration example for one-to-one binding between FXS
and FXO
Network requirements
•
Router A and Router B are connected over an IP network and a PSTN. Telephone A attached to
Router A can make calls to Telephone B attached to Router B over the IP network or the PSTN.
•
Usually, Telephone A makes calls to Telephone B over the IP network. In the case that the IP
network is unavailable, Router A sends call from Telephone A through the bound FXO interface
to Telephone B over PSTN.
Figure 15 Network diagram
Configuration outline
•
Configure one-to-one binding between FXS and FXO voice subscriber lines.
•
When the IP network is available, the VoIP entity is preferably used to make calls over the IP
network.
55
•
When the IP network is unavailable, the POTS entity is used to make calls through the bound
FXO voice subscriber line over the PSTN.
Configuration procedure
Routing-related configurations are beyond the scope of this example. This example assumes that
Router A and Router B are reachable to each other.
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
# Configure a VoIP entity for IP calls and set the match template to 210….
[RouterA-voice-dial] entity 210 voip
[RouterA-voice-dial-entity210] match-template 210....
[RouterA-voice-dial-entity210] address sip ip 192.168.0.76
[RouterA-voice-dial-entity210] quit
# Configure a POTS entity for the FXS voice subscriber line.
[RouterA-voice-dial] entity 0101001 pots
[RouterA-voice-dial-entity101001] match-template 0101001
[RouterA-voice-dial-entity101001] line 6/24
[RouterA-voice-dial-entity101001] quit
# Configure a backup POTS entity on the FXO voice subscriber line, set the match template
to .T, enable the sending of all digits of a called number, and configure a calling number
permitted to originate calls to the POTS entity.
[RouterA-voice-dial] entity 211 pots
[RouterA-voice-dial-entity211] match-template .T
[RouterA-voice-dial-entity211] line 6/0
[RouterA-voice-dial-entity211] send-number all
[RouterA-voice-dial-entity211] caller-permit 0101001
[RouterA-voice-dial-entity211] quit
[RouterA-voice-dial] quit
[RouterA-voice] quit
# Configure the PLAR function and the delay off-hook binding.
[RouterA] subscriber-line 6/0
[RouterA-subscriber-line6/0] private-line 0101001
[RouterA-subscriber-line6/0] hookoff-mode delay bind 6/24
[RouterA-subscriber-line6/0] quit
# Configure a dial plan where the VoIP entity is preferred.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule type-first 2 1 3 4
2.
Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
# Configure a VoIP entity for IP calls and set the match template to 010….
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 192.168.0.71
[RouterB-voice-dial-entity10] quit
56
# Configure a POTS entity on the FXS voice subscriber line.
[[RouterB-voice-dial] entity 2101002 pots
[RouterB-voice-dial-entity2101002] match-template 2101002
[RouterB-voice-dial-entity2101002] line 6/24
[RouterB-voice-dial-entity2101002] quit
# Configure a backup POTS entity on the FXO voice subscriber line, set the match template
to .T, enable the sending of all digits of a called number, and configure a calling number
permitted to originate calls to the POTS entity.
[RouterB-voice-dial] entity 011 pots
[RouterB-voice-dial-entity11] match-template .T
[RouterB-voice-dial-entity11] line 6/0
[RouterB-voice-dial-entity11] send-number all
[RouterB-voice-dial-entity11] caller-permit 2101002
[RouterB-voice-dial-entity11] quit
[RouterB-voice-dial] quit
[RouterB-voice] quit
# Configure the PLAR function and the delay off-hook binding.
[RouterB] subscriber-line 6/0
[RouterB-line6/0] private-line 2101002
[RouterB-line6/0] hookoff-mode delay bind 6/24
[RouterB-line6/0] quit
# Configure a dial plan where the VoIP entity is preferred.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] select-rule type-first 2 1 3 4
3.
In the case that the IP network is unavailable, configure Router A and Router B as follows:
# Configure Router A, with the POTS entity preferred and the other configurations remaining
unchanged.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule type-first 1 2 3 4
# Configure Router B, with the POTS preferred and the other configurations remaining
unchanged.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] select-rule type-first 1 2 3 4
Troubleshooting analog voice subscriber line
configuration
Failed to hang up
Symptom
The FXO voice subscriber line cannot detect busy tone signals sent from the PBX, so the line is in
connection even if the remote end hangs up.
57
Figure 16 Network diagram
As shown in Figure 16, suppose Telephone A hangs up first after conversation, then PBX A plays
busy tones to Router A, which disconnects the line after detecting the busy tones and sends a
disconnect message to Router B. Router B sends the message to PBX B that then plays busy tones
to Telephone B. The whole disconnection process is done. If the FXO voice subscriber line on Router
A cannot detect the busy tone played by PBX A, the call cannot be terminated.
Solution
•
If the PBX uses North American busy tone specification but the router uses European
specification (the default specification), issue the area north-america command in subscriber
line view on the router to change the busy tone specification. If the problem persists, proceed
with the next step.
•
Automatic busy tone recognition: The automatic busy tone recognition technology uses
software to sample and analyze busy tones to produce a set of parameters that represent the
most common patterns of busy tones. After configured with these parameters, the FXO ports
can recognize different types of busy tones.
For information about the procedure of automatic busy tone recognition test, see "Busy tone
detection." If the test fails, repeat the operations till the busy tones can be recognized.
58
Configuring digital voice subscriber lines
This chapter covers the configuration of E1, T1, and BSV voice subscriber lines.
Introduction to E1 and T1
Overview
Plesiochronous digital hierarchy (PDH) includes two major communications systems: ITU-T E1
system and ANSI T1 system. The E1 system is dominant in Europe. The T1 system is dominant in
USA, Canada and Japan.
E1 and T1 use the same sampling frequency (8 kHz), PCM frame length (125 μs), bits per code word
(8 bits) and timeslot bit rate (64 kbps). They differ in these aspects:
•
E1 adopts A law coding/decoding of 13-segment but T1 adopts μ law coding/decoding of
15-segment.
•
Each PCM primary frame of E1 contains 32 timeslots but that of T1 contains 24 timeslots. Each
PCM primary frame of E1 contains 256 bits but that of T1 contains 193 bits. Therefore, E1
provides 2.048 Mbps bandwidth and T1 provides 1.544 Mbps bandwidth.
E1 and T1 voice functions
E1 and T1 mainly provide voice and signaling trunks to the PSTN. To realize this function, the router
must have E1 and T1 voice interfaces and be configured with functions required for transmitting
voice over E1 and T1 lines.
The E1 and T1 voice physical interfaces are VE1 and VT1 interfaces, respectively.
PSTN and routers are connected through E1/T1 trunks, as shown in Figure 17.
Figure 17 Network diagram
E1/T1 voice transmission allows a router to provide more channels of voice communication, greatly
improving router utilization and broadening service range.
E1 and T1 interfaces
This section covers information related to E1 and T1 interfaces.
E1 interface
An E1 interface is logically divided into timeslots (TSs) with TS 16 being a signaling channel.
On E1 interfaces, you can create PRI groups or TS sets.
59
You can use an E1 interface as an ISDN PRI or CE1 interface:
•
As an ISDN PRI interface, the E1 interface adopts DSS1 or QSIG signaling. As TS 0 is used to
transfer synchronization information and TS16 is used as a D channel to transfer connection
signaling, you can arbitrarily bind any timeslots other than TS0 and TS16 as a logical interface,
which is equivalent to an ISDN PRI interface.
•
As a CE1 interface with a signaling channel, the E1 interface can adopt R2 signaling, digital
E&M signaling, or digital LGS signaling.
{
{
When R2 signaling is adopted, every 32 timeslots form a primary frame (PCM30 for
example), where TS0 is used for frame synchronization, TS16 for digital line signaling, and
other 30 timeslots for voice transmission. Every 16 primary frames form one multiframe. In
each multiframe, TS0 in even primary frames conveys frame alignment signal (FAS) and
TS0 in odd primary frames conveys non–FAS (NFAS) about link status information. NFAS
provides control signaling for primary rate multiplexing. In the first primary frame, frame 0,
the high-order 4 bits in TS16 convey multiframe FAS (MFAS) and the lower-order 4 bits
convey non-multiframe FAS (NMFAS). TS16 in each of other 15 primary frames conveys
line status information for two timeslots. For example, TS16 in frame 1 conveys the digital
line signaling status of TS1 and TS17 while that in frame 2 conveys the digital line signaling
status of TS2 and TS18, and so on.
When digital E&M signaling is adopted, the E1 interface functions as a digital E&M interface.
On the interface, timeslot division and functions are the same as those with R2 signaling.
When digital LGS signaling is adopted, the E1 interface functions as a digital FXO or FXS interface.
On the interface, timeslot division and functions are the same as those with R2 signaling.
After you create a TS set and configure signaling on an E1 voice interface card, the system can
automatically create the voice subscriber line for the TS set.
After TSs of an E1 interface are bound to form a PRI group, the system will automatically generate
the corresponding voice subscriber line.
T1 interface
A T1 interface can be physically divided into 24 timeslots numbered TS1 through TS24.
You can use a T1 interface as an ISDN PRI interface. The interface adopts DSS1 or QSIG signaling.
Except TS24, which is used as D channel for signaling, you can arbitrarily bundle other timeslots into
an interface logically equivalent to an ISDN PRI interface.
In addition to DSS1 and QSIG signaling, T1 interfaces support R2 signaling, digital E&M signaling,
and LGS signaling. Configured with digital E&M signaling, a T1 interface is used as a digital E&M
interface; with digital LGS signaling, a digital FXO or FXS interface.
Like E1 voice interface cards, T1 voice interface cards also have the features of voice subscriber
lines.
Features of E1 and T1
E1 and T1 are characterized by the following:
•
Signaling modes
•
Fax function
•
Protocols and standards
Signaling modes
E1/T1 interfaces support these types of signaling:
•
DSS1/QSIG user signaling—Adopted on the D channel between ISDN user and network
interface (UNI). It comprises a data link layer protocol and a Layer 3 protocol used for basic call
control.
60
•
ITU-T R2 signaling—Includes digital line signaling and interregister signaling. Digital line
signaling is transmitted in TS16 (ABCD bits) of E1 trunk. It conveys status information about E1
trunks to describe whether the trunks are occupied, released, or blocked. Interregister signaling
conveys information about address, language and discriminating digits for internal calls, echo
suppressor, caller properties and callee properties in multi-frequency compelled approach
(forward and backward) in each timeslot.
•
Digital E&M signaling—Similar to R2 signaling. It transmits E (recEive) and M (transMit) call
control signals similar to analog E&M signaling in TS16, alignment signals in TS0, and voice
signals in other timeslots. In digital E&M signaling, when an E1 trunk detects and sends
connection signaling, it looks at the signal in TS16. Digital E&M signaling provides three start
modes, immediate, wink, and delay, to adapt to different devices for more reliable connection.
•
Digital LGS—Digital loop start signaling is used between telephones and switches to identify
the off-hook/on-hook state, while ground-start signaling is used between switches. They differ
in that the two parties in conversation must check grounding state before closing the line in the
ground-start approach.
Fax function
The FAX function is available on E1/T1 voice interfaces to set up fax channels and transmit/receive
fax data.
Protocols and standards
E1/T1 voice supports SIP and recommendations in the ITU-T H.323 framework, and G.711, G.729,
and G.723.1 Annex A (5.3K and 6.3K) in ITU standards.
Table 4 Protocols supported by E1/T1
Item
E1 Voice
T1 Voice
Framing format
Cyclic redundancy check 4 (CRC4),
non-CRC4
Super frame (SF), extended super
frame (ESF)
Line coding format
High-density bipolar 3 (HDB3), alternate
mark inversion (AMI)
Bipolar 8-zero substitution (B8ZS), AMI
Hardware compatibility with digital voice
subscriber lines
Digital voice subscriber lines are not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
E1 and T1 configuration task list
Configuration considerations:
61
1.
Install a VE1 or VT1 voice interface card on the router.
2.
Configure basic parameters for voice interfaces.
3.
Configure the corresponding voice subscriber line for each TS set.
4.
Configure signaling.
Task
Remarks
Configuring basic parameters for an E1 voice interface
Optional.
Configuring basic parameters for a T1 voice interface
Optional.
Configuring the voice subscriber line for a TS set
Required.
Binding logical voice subscriber line to POTS entity
Required.
Configuring R2 signaling
Configuring basic R2 signaling parameters
Optional.
Configuring R2 digital line signaling
Optional.
Configuring R2 interregister signaling
Optional.
Configuring PRI
Required.
Configuring digital E&M signaling
Optional.
Configuring digital LGS signaling
Optional.
Configuring AMD
Optional.
Configuring reverse charging function
Optional.
Enabling the router to treat DISCONNECT messages with PI 8 as standard
DISCONNECT messages
Optional.
Mirroring PCM, RTP packets, or voice command data on a digital voice
subscriber line
Optional.
Configuring basic parameters for an E1 voice
interface
This section describes the basic parameters and procedures for configuring an E1 voice interface.
Configuring a TDM clock source
Introduction to TDM clock source
When digital voice E1 interfaces perform time-division multiplexing (TDM) timeslot interchange, it is
important for them to achieve clock synchronization to prevent frame slips and bit errors.
Depending on your configurations on E1 interfaces, the system adopts different clocking approaches.
When there is a voice co-processing module (VCPM) on the main board, the clock distribution
principle is as follows:
•
If the clock mode is set to line for all interfaces, the clock on the interface with the lowest
number is adopted. If the interface goes down, the clock on the interface with the next lowest
number is adopted.
•
If the clock mode is set to line primary for one interface, the clock on the interface is adopted.
In one system, you can do this on only one interface.
•
If the line keyword is specified for one interface and the internal keyword for all others, the
clock on the interface is adopted.
62
•
Generally, you cannot set the clock source for all interfaces in a system to internal. This is to
prevent frame slips and bit errors. You can do this, however, if the remote E1 interfaces adopt
the line clock source.
When there is no VCPM on the main board, the configuration of each MIM/FIC is independent, but
the clock mode of only one interface can be set to line primary on a device.
Suppose that you remove the FIC on an interface whose clock mode is set to line primary without
powering it off and that you set the clock mode to line primary for another interface. In this case, if
you re-insert the FIC, the clock mode of the FIC is restored to internal, instead of remaining line
primary. In this way, there is only one interface whose clock mode is set to line primary on a device.
For example, after the FIC in slot 5 (interface 5/0) whose clock mode is set to line primary is
removed without being powered off and the clock mode of interface 6/0 is set to line primary, the
clock mode of interface 5/0 will be restored to internal, instead of being line primary, if the FIC is
re-inserted.
NOTE:
The line clock source and internal clock source are referred to as slave and master clock modes in
some features.
Configure a TDM clock source
To configure a TDM clock source for E1 interfaces:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 slot-number
N/A
Optional.
3.
Configure a TDM clock
source for the E1 interface.
tdm-clock { internal | line
[ primary ] }
By default, the TDM source clock
for an E1POS interface is line
TDM clock, and the TDM clock
source for other E1 interfaces is
the internal clock.
Configuring the framing format and line coding format
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 slot-number
N/A
3.
Configure the framing
format.
frame-format { crc4 | no-crc4 }
Configure the line coding
format.
code { ami | hdb3 }
4.
Optional.
The default is non-CRC4.
Optional.
The default is HDB3.
For more information about the controller e1, frame-format, and code commands, see Interface
Command Reference.
63
Creating a TS set
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 slot-number
N/A
3.
Create a TS set according to
the selected signaling mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink | fxo-ground |
fxo-loop | fxs-ground | fxs-loop |
r2 }
By default, no TS set is
configured.
4.
Enter CAS view.
cas ts-set-number
N/A
5.
Configure an E1 trunk
routing mode.
select-mode { max | maxpoll |
min | minpoll }
Optional.
By default, the timeslot with the
lowest number is selected from all
available timeslots for routing.
Set the physical state change suppression interval on an E1
interface
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 slot-number
N/A
3.
Set the physical state
change suppression interval
on the E1 interface.
Optional.
link-delay delay-time
By default, the physical state
change suppression is disabled
on an E1 interface.
Restoring default settings for an E1 voice interface
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 slot-number
N/A
3.
Restore the default settings
on the E1 voice interface.
default
Optional.
Configuring basic parameters for a T1 voice
interface
This section describes the basic parameters and procedures for configuring a T1 voice interface.
64
Configuring a TDM clock source
The TDM clock source configuration for a T1 interface is similar to that for an E1 interface. For more
information, see "Introduction to TDM clock source."
To configure a TDM clock source:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller t1 slot-number
N/A
3.
Configure a TDM clock
source for the T1 interface.
tdm-clock { internal | line
[ primary ] }
Optional.
By default, the internal clock is
used as the TDM clock source.
Configuring the framing format and line coding format
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller t1 slot-number
N/A
3.
Configure the framing
format.
frame-format { esf | sf }
Configure the line coding
format.
code { ami | b8zs }
4.
Optional.
The default is ESF.
Optional.
The default is B8ZS.
For more information about the frame-format and code commands, see Interface Command
Reference.
Creating a TS set
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller t1 slot-number
N/A
3.
Create a TS set
according to the selected
signaling mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink | fxo-ground | fxo-loop |
fxs-ground | fxs-loop | r2 }
By default, no TS set is
configured.
4.
Enter CAS view.
cas ts-set-number
N/A
5.
Configure a T1 trunk
routing mode.
select-mode [ max | maxpoll | min |
minpoll ]
Optional.
65
By default, the timeslot with the
lowest number is selected from all
available timeslots for routing.
Set a physical state change suppression interval on a T1
interface
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller T1 slot-number
N/A
3.
Set a physical state change
suppression interval on the
T1 interface.
Optional.
link-delay delay-time
By default, the physical state
change suppression is disabled
on a T1 interface.
Restoring default settings for a T1 voice interface
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller t1 slot-number
N/A
3.
Restore the default settings
on the T1 voice interface.
default
Optional.
Configuring the voice subscriber line for a TS set
A TS set is a list of timeslots on an E1/T1 interface which abstractly forms a logical voice subscriber
line used for the configuration of R2 signaling, digital E&M signaling, digital LGS signaling and other
voice functions. For each TS set, the system automatically creates a logical voice subscriber line
numbered in the form of E1/T1 interface number:TS set number. On the voice subscriber line, you
can conveniently configure signaling and other voice functions for the corresponding E1/T1 line.
Note that on each E1/T1 interface you can create only one TS set.
The digital subscriber lines have the same functions as analog subscriber lines do. For more
information, see "Configuring analog voice subscriber lines."
For the description, shutdown, private-line, receive gain, transmit gain, echo-canceller,
nlp-on, and cng-on commands, see Voice Command Reference.
Configuration prerequisites
Complete basic parameters configuration for the VE1/VT1 interface you are working with.
Configuring basic functions for the voice subscriber line
To configure the basic functions for the voice subscriber line:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } number
N/A
3.
Create a TS set and define
its signaling type.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
N/A
66
Step
Command
Remarks
{ e&m-delay | e&m-immediate |
e&m-wink | r2 | fxo-ground |
fxs-loop | fxs-ground | fxo-loop |
r2 }
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line
slot-number:ts-set-number
N/A
6.
Configure a companding law
for signal quantization.
7.
Create a description for the
voice subscriber line.
description text
8.
Shut down the voice
subscriber line.
shutdown
Optional.
pcm { a-law | μ-law }
A-law for a VE1 interface card and
μ-law for a VT1 interface card by
default.
Optional.
The description for a voice
subscriber line is
interface-name+Interface.
Optional.
The voice subscriber line is up by
default.
Configuring the DTMF detection sensitivity
Use the dtmf threshold digital command to configure the DTMF detection sensitivity. A high DTMF
detection sensitivity indicates a high tolerance of DTMF digits collection, a high probability of
detection errors, and a low probability of DTMF failing to be detected.
To configure a DTMF detection sensitivity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface
view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set
according to the
selected signaling
mode.
timeslot-set ts-set-number timeslot-list
timeslots-list signal { e&m-delay |
e&m-immediate | e&m-wink |
fxo-ground | fxo-loop | fxs-ground |
fxs-loop | r2 }
N/A
4.
Exit E1/T1 interface
view.
quit
N/A
5.
Enter voice
subscriber line view.
subscriber-line
slot-number:ts-set-number
N/A
Optional.
6.
Configure a DTMF
detection sensitivity.
dtmf threshold digital value
67
By default, the DTMF detection
sensitivity level is 0, that is,
insensitive.
This command is applicable to the
BSV voice subscriber line, and is
inapplicable to E1 and T1 voice
subscriber lines.
Configuring the volume adjustment function
To configure the volume adjustment function:
Step
Command
Remarks…
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set according to
the selected signaling mode.
timeslot-set ts-set-number timeslot-list
timeslots-list signal { e&m-delay |
e&m-immediate | e&m-wink | fxo-ground
| fxo-loop | fxs-ground | fxs-loop | r2 }
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line slot-number:ts-set-number
N/A
6.
Configure the input gain on
the voice interface.
receive gain value
Configure the output gain on
the voice interface.
transmit gain value
7.
Optional.
The default is 0.
Optional.
The default is 0.
IMPORTANT:
Gain adjustment might lead to a call failure. H3C recommends not adjusting the gain. If necessary,
make sure you understand the impact of the adjustment on your call before you adjusting the gain.
Configuring the echo adjustment function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set according to
the selected signaling mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink | fxo-ground |
fxo-loop | fxs-ground | fxs-loop |
r2 }
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line
slot-number:ts-set-number
N/A
6.
Enable the echo cancellation
function.
echo-canceller enable
Enabled by default.
7.
Configure echo duration.
echo-canceller tail-length
milliseconds
Optional.
The default is 0 milliseconds.
Optional.
8.
Enable the nonlinear function
of echo cancellation.
Enabled by default.
nlp-on
This command takes effect only
after the echo-canceller enable
command is issued.
68
Configuring the comfortable noise function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set according to
the selected signaling mode.
timeslot-set ts-set-number timeslot-list
timeslots-list signal { e&m-delay |
e&m-immediate | e&m-wink | fxo-ground
| fxo-loop | fxs-ground | fxs-loop | r2 }
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line slot-number:ts-set-number
N/A
6.
Enable the comfortable noise
function.
cng-on
Optional.
Enabled by default.
Configuring options related to dial plan
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set according to
the selected signaling mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink | fxo-ground |
fxo-loop | fxs-ground | fxs-loop |
r2 }
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line
slot-number:ts-set-number
N/A
6.
Enable the private line auto
ring function.
private-line string
Bind a calling/called number
substitution rule list to a voice
subscriber line.
substitute { called | calling }
list-number
7.
Optional.
Disabled by default.
By default, no calling/called
number substitution rule list is
bound to a voice subscriber line.
For more information about the above commands, see Voice Command Reference.
Binding logical voice subscriber line to POTS
entity
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set according to
timeslot-set ts-set-number
N/A
69
Step
Command
Remarks
the selected signaling mode.
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink | fxo-ground |
fxo-loop | fxs-ground | fxs-loop |
r2 }
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter voice view.
voice-setup
N/A
6.
Enter voice dial program
view.
dial-program
N/A
7.
Create a POTS entity and
enter POTS entity view.
entity entity-number pots
N/A
8.
Bind a logical voice
subscriber line to a POTS
entity.
Optional.
line slot-number:ts-set-number
By default, no logical voice
subscriber line is bound to any
POTS entity.
Configuring R2 signaling
Introduction to R2 signaling
ITU-T recommendations Q.400 through Q.490 define the R2 signaling standards. However, the R2
signaling standards implemented in different countries and regions are ITU variants.
R2 signaling includes two categories: digital line signaling and interregister signaling. Digital line
signaling conveys status information about E1 trunks to describe whether the trunks are occupied,
released, or blocked. Interregister signaling transmits such information as address in multi-frequency
compelled approach. Generally, the calling side serves as the originating PBX and the called side
serves as the terminating PBX. Signals sent by the originating PBX are called forward signals and
those sent by the terminating PBX are called backward signals, as shown in Figure 18.
Figure 18 R2 signaling elements
ITU-T digital line signaling
Digital line signaling is responsible for changing call statuses and conditions of a line. It functions to
identify and detect these four states: calling party goes off-hook and seizes the line, called party goes
off-hook and answers the call, calling party releases the call, and called party releases the call.
Accordingly, it sets the line to be idle or seized. This signaling is transmitted in the 16th multiframe
timeslot of PCM system. The two transmission directions of each line have 4 bits (A, B, C and D) as
flag bits, with C and D bits fixed to 01. Therefore, the forward line signaling adopts af and bf bits and
the backward line signaling adopts ab and bb bits, as shown in the following table:
70
Table 5 Line signaling bit description
Bit
Description
Vale = 0
Value = 1
af
Identifies working state of device at the
originating side and indicates state of the calling
party line.
Off-hook, seized
On-hook (idle)
bf
Indicates fault state from the originating side to
the terminating side.
Normal
Faulty
ab
Indicates state of the called party line (on-hook
or off-hook).
Off-hook by called
party
On-hook by called party
bb
Indicates state of device at the terminating side
(idle or seized).
Idle
Seized or blocked
Table 6 State code of line signaling
Signaling code
State of the circuit
Forward
Backward
af
bf
ab
bb
Idle
1
0
1
0
Seized
0
0
1
0
Seizure-ack
0
0
1
1
Answer
0
0
0
1
Clear-back
0
0
1
1
Clear-forward
0
0
0/1
1
Blocked
1
0
1
1
Unblocked
1
0
1
0
The following are typical R2 digital line signaling interaction procedures:
1.
Call establishment.
When the circuit is idle, the originating side sends a forward seizure signal to the terminating
side. The terminating side then sends back a seizure acknowledgement signal after it
recognizes the seizure signal. At this time, the circuits of the both sides are seized, and they
start interregister signaling exchange. When the called party picks up the phone, the
terminating side sends a backward answer signal. After the originating side recognizes the
received signal, it establishes the call.
Figure 19 R2 digital line signaling – call establishment
2.
Originating side releases the call.
71
The originating side sends a clear-forward signal 10. When the terminating side recognizes the
clear-forward signal, it sends a backward signal 10 (release guard signal or clear-forward
acknowledgement signal). After the originating side recognizes the backward signal 10, it
releases the circuit.
Figure 20 R2 digital line signaling – originating side releases the call
3.
Terminating side releases the call.
The terminating side sends a clear-back signal 11. After the originating side receives the
clear-back signal, it sends a clear-forward signal 10. After the terminating side recognizes the
forward signal 10 sent by the originating side, it sends a backward signal 10. After the
originating side recognizes the backward signal 10, it releases the circuit.
Figure 21 R2 digital line signaling – terminating side releases the call
4.
Line released by forced release signal.
When the terminating side supports metering signals, the system might send a forced release
signal 00 instead of a clear-back signal 11 to release the line. This is to avoid collision between
the clear-back signal sent by the called party and the metering signal.
5.
Blocking in idle state or during conversation.
After the originating side receives a blocking signal 11 from the terminating side when the circuit
is idle or during conversation, it sends a forward signal 10. At this time, the circuit is blocked.
When the terminating side unblocks the circuit, it sends a backward signal 10 in the
corresponding line to indicate that the line is idle. The originating side should maintain the
forward signal 10 and unblock the local-end circuit for next call.
6.
Troubleshooting in idle state.
If the terminating side receives a forward signal 11 from the originating side to indicate a device
fault when the circuit is idle, the terminating side sends a backward signal 11. Then, the circuit is
in faulty state. When the device recovers, the originating side sends the forward signal 10, and
the terminating side responds with the signal 10. At this time the circuit regains normal state.
7.
Troubleshooting during conversation.
After the terminating side receives a forward signal 11 from the originating side to indicate
device fault during conversation, the terminating side releases the line backward. At the same
time, it sends a backward signal 11. Then, the circuit is in faulty state. When the device recovers,
72
the originating side sends a forward signal 10, and the terminating side sends back a signal 10.
At this time, the circuit recovers.
ITU-T interregister signaling
Interregister signaling controls automatic connection of circuits. It adopts MFC mode and includes
forward signaling and backward signaling. Forward signaling exchange includes Group I and Group
II, while backward signaling exchange includes Group A and Group B. When the originating side
recognizes the seizure acknowledgement signal, the register begins to send the first digit of the
called number, and waits for the response of Group A signaling from the terminating side.
•
Group I forward signals—Include connection control signals and digit signals.
Table 7 Forward Group I signals
Designation
Basic meaning
I-1 through I-10
Digits 1, 2, 3, 4, 5, 6, 7, 8, 9, and 0, responsible for sending number
information to the terminating side
I-11
Spare for national use
I-12
Request refused
I-13
Connected to tested device
I-14
Spare for national use
I-15
Address identification terminator and pulse terminator (used in international
calls)
•
Group A backward signals—Control signals used for controlling and acknowledging Group I
forward signals.
Table 8 Group A backward signals
•
Designation
Basic Meaning
A-1
Send next digit
A-2
Send last but one digit
A-3
Address-complete; changeover to reception of Group B signals
A-4
Congestion in the national network; terminate interregister signaling exchange
A-5
Send calling party’s category
A-6
Address-complete; terminate interregister signaling exchange, charge, and set up
speech conditions
A-7
Send last but two digits
A-8
Send last but three digits
A-9
Spare for national use
A-10
Spare for national use
A-11
Send country code indicator
A-12
Send language or discrimination digit
A-13
Send nature of circuit
A-14
Request for information on use of an echo canceller
A-15
Congestion in an international exchange; terminate interregister signaling interaction
Group II forward signals—Identify the calling party category. The system looks at the calling
party category to decide whether the calling party can perform forced release or break-in.
73
Table 9 Group II forward signals
Designation
Basic Meaning
II-1
Subscriber without priority
II-2
Subscriber with priority
II-3
Maintenance equipment
II-4
Spare for national use
II-5
Operator
II-6
Data transmission
II-7
Subscriber (or operator without forward transfer facility), for international use
II-8
Data transmission (for international use)
II-9
Subscriber with priority (for international use)
II-10
Operator with forward transfer facility (for international aid use)
II-11 through II-15
Spare for national use
•
Group B backward signals—Identify the state of the called party, and acknowledge Group II
signals and control connection.
Table 10 Group B backward signals
Designation
Basic Meaning
B-1
Spare for national use
B-2
Send special information tone
B-3
Subscriber line busy
B-4
Congestion
B-5
Unallocated number
B-6
Subscriber line free, charge
B-7
Subscriber line free, no charge
B-8
Subscriber line out of order
B-9 through B-15
Spare for national use
The following figure shows the exchange process requesting calling party information, which is
typical of R2 interregister signaling.
74
Figure 22 ITU-T R2 interregister signaling exchange process
Originating side
Calling number: 123
Terminating side
Called number 789
Line signaling exchange
Send called number digit 7 (I-7)
Request next digit (A-1)
Send called number digit 8 (I-8)
Request calling party information (A-5)
Send calling accounting category 2 (II-7)
Request calling party information (A-5)
Send calling number digit 1 (I-1)
Request calling party information (A-5)
Send calling number digit 2 (I-2)
Request calling party information (A-5)
Send calling number digit 3 (I-3)
Request calling party information (A-5)
Send number terminator (I-15)
Request next digit (A-1)
Send called number digit 8 (I-9)
All digits are received completely (A-3)
Send calling service category 2 (II-2)
Subscriber idle and charging starts (B-6)
Line signaling exchange
Configuring basic R2 signaling parameters
Configuring the country or region mode
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Configure the country or
region mode.
mode zone-name
[ default-standard ]
Optional.
By default, the ITU-T mode is
adopted.
Configuring the trunk direction for R2 signaling
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
timeslot-set ts-set-number
N/A
75
Step
Command
Remarks
R2 signaling for it.
timeslot-list timeslots-list signal
r2
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Configure the trunk direction
for R2 signaling.
trunk-direction timeslots
timeslots-list { dual | in | out }
Optional.
Bidirectional by default.
Enabling the terminating side to send busy tones to the calling subscriber
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Enable the terminating side
to send busy tones to the
calling subscriber.
sendring ringbusy enable
Configure the duration of
playing busy tones.
timer ring { ringback |
ringbusy } time
6.
Optional.
By default, the terminating side
sends busy tones to the calling
subscriber.
Optional.
By default, the duration of playing
busy tones is 30,000 milliseconds.
Enabling the DTMF mode to receive and send R2 signaling
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Enable the receiving and
sending of R2 signaling in
the DTMF mode.
Optional.
dtmf enable
By default, R2 signaling is
received and sent in the
multi-frequency compelled (MFC)
mode.
Optional.
6.
Configure the delay before
sending DTMF signals.
The default is 50 milliseconds.
timer dtmf time
You must configure the dtmf
enable command before this
command.
Configuring the connection mode for an R2 call
There are two connection modes for an R2 call: terminal-to-terminal (terminal) and
segment-to-segment (segment) mode.
76
•
In the terminal-to-terminal (terminal) mode, after the called number is received, the R2 protocol
module must wait for the real state (busy or idle) of the called party before returning the
corresponding register information to the originating side.
•
In the segment-to-segment (segment) mode, after the called number is received, the R2
protocol module directly returns the "called party idle" register signaling, without waiting for the
real state of the terminating side.
To configure the connection mode for an R2 call:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Configure the connection
mode for an R2 call.
Optional.
callmode { terminal | segment }
By default, the
terminal-to-terminal
(terminal) mode is adopted.
Maintaining the circuits of a timeslot or a range of timeslots
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number timeslot-list
timeslots-list signal r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Maintain the circuits of a
timeslot or a range of
timeslots.
ts { block | open | query | reset } timeslots
timeslots-list
Optional.
Setting the length of called numbers that can be received
With an E1POS card installed, a point of sale (POS) terminal can access the network using R2
signaling. You can accelerate signaling exchange by setting the length of called numbers received
by the E1POS card. This will meet the real time transmission requirements of the POS terminal
access service.
To set the length of called numbers that can be received:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number timeslot-list
timeslots-list signal r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Set the length of called
numbers that can be
received.
Optional.
posa called-length calledlength
77
By default, the length
of the called numbers
that can be received
Step
Command
Remarks
is 31 digits.
For more information about the POS terminal service, see Terminal Access Configuration Guide.
Configuring R2 digital line signaling
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Enable the terminating side
to send answer signal.
answer enable
6.
7.
8.
9.
Optional.
Enabled by default.
Optional.
Enable the originating side to
support re-answer signal
processing.
re-answer enable
Enable the terminating side
to send clear-forward
acknowledgement signal
(clear-back signal).
clear-forward-ack enable
Enable metering for R2
signaling.
force-metering enable
Enable the terminating side
to send seizure
acknowledgement signal.
seizure-ack enable
10. Configure the ABCD bit
pattern for each type of R2
line signal.
dl-bits { answer | blocking |
clear-back | clear-forward | idle |
seize | seizure-ack |
release-guard } { receive |
transmit } ABCD
11. Configure the C and D signal
bits.
renew ABCD
12. Configure line signal
inversion mode.
reverse ABCD
By default, the originating side
does not support processing of
re-answer signals.
Optional.
Disabled by default.
Optional.
Disabled by default.
Optional.
Enabled by default.
Optional.
For more information, see Table
11.
Optional.
N/A
Optional.
The default is 0000, that is, line
signal inversion disabled.
Optional.
13. Set timeout values of line
signals.
timer dl { answer | clear-back |
clear-forward | | re-answer |
release-guard | seize } time
78
The default is 60,000 milliseconds
for answer signal; 10,000
milliseconds for clear-back signal
and clear-forward signal and
release-guard signal; 1,000
milliseconds for seizure signal
and re-answer signal.
Table 11 Default values of signals in R2 digital line signaling
Signal
Default rx-bits ABCD
Default tx-bits ABCD
Answer
0101
0101
Blocking
1101
1101
Clear-back
1101
1101
Clear-forward
1001
1001
Idle
1001
1001
Seize
0001
0001
Seizure-ack
1101
1101
Release-guard
1001
1001
Configuring R2 interregister signaling
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set and enable
R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
r2
N/A
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Enable the terminating side
to request calling party
information.
ani { all | ka }
6.
Set the range for the number
of calling number digits to be
collected.
ani-collected min min-value max
max-value
7.
Configure the number of
digits that should be
collected to request a calling
party flag.
ani-offset number
Set the interdigit timeout time
in interregister signaling.
ani-timeout timer-length
Adopt Group B signals to
complete interregister
signaling exchange.
group-b enable
8.
9.
Optional.
Disabled by default.
Optional.
By default, no restriction is placed
on the number of calling number
digits to be collected.
Optional.
The default for the number
argument is 1.
Optional.
3 seconds by default.
Optional.
10. Enable this end to send a
number terminator to the
terminating side.
final-callednum enable
11. Configure the special
characters that are
supported during
interregister signaling
exchange.
special-character character
number
12. Configure register signal
values in R2 signaling.
register-value { billingcategory |
callcreate-in-groupa |
79
Group B signals are used by
default.
Optional.
Disabled by default.
Optional.
No special character is configured
by default.
Optional.
The defaults vary by country
Step
Command
Remarks
callingcategory | congestion |
demand-refused | digit-end |
nullnum | req-billingcategory |
req-callednum-and-switchgrou
pa | req-callingcategory | reqcurrentcallednum-in-groupc |
req-currentdigit | reqfirstcallednum-in-groupc |
req-firstcallingnum |
req-firstdigit | req-lastfirstdigit |
req-lastseconddigit |
req-lastthirddigit |
req-nextcallednum |
req-nextcallingnum |
req-switch-groupb |
subscriber-abnormal
|subscriber-busy |
subscriber-charge
|subscriber-idle } value
mode.
13. Configure the duration of
register pulse signals such
as A-3, A-4, and A-6 in R2
signaling.
timer register-pulse
persistence time
Optional.
14. Configure the maximum time
waiting for a Group B signal.
timer register-complete
group-b time
The default is 150 milliseconds.
Optional.
The default is 30,000
milliseconds.
Configuring PRI
This section describes how to configure DSS1 and QSIG signaling and how to enable the
transmission of QSIG signaling over a SIP network.
Configuring DSS1 and QSIG signaling
After you create a PRI group with the pri-set command on an E1/T1 interface, a serial interface is
automatically created. This interface is named serial number:15 on an E1 interface and serial
number:23 on a T1 interface. You can enter the view of the interface with the interface serial
number:{ 15 | 23 } command to configure DSS1 and QSIG user signaling.
To configure DSS1 or QSIG signaling:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Bundle timeslots into a PRI
group.
pri-set [ timeslot-list range ]
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter the view of the serial
interface created for the PRI
group.
interface serial number: { 15 |
23 }
N/A
Enable DSS1 or QSIG
signaling.
isdn protocol-type { dss1 |
qsig }
By default, DSS1 signaling is
enabled.
6.
80
Step
Command
Remarks
7.
isdn protocol-mode { network |
user }
The default is user.
Configure a signaling
protocol mode.
For more information about the isdn protocol-type and isdn protocol-mode commands, see Layer
2—WAN Command Reference.
Enabling the transmission of QSIG signaling over a SIP
network
With the SIP for telephones (SIP-T) protocol, the ISDN signaling messages can be tunneled through
the SIP messages. The device only supports encapsulating QSIG messages within SIP messages.
For more information about the SIP-T protocol, see "Configuring SIP."
To configure the SIP tunneling function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Bundle timeslots into a PRI
group.
pri-set [ timeslot-list range ]
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter the view of the serial
interface created for the PRI
group.
interface serial number: { 15 |
23 }
N/A
6.
Enable QSIG signaling.
isdn protocol-type qsig
By default, DSS1 signaling is
enabled.
7.
Exit E1/T1 interface view.
quit
N/A
8.
Enter the view of the voice
subscriber line of the PRI
group.
subscriber-line number: { 15 |
23 }
N/A
Enable the QSIG tunneling
function.
qsig-tunnel enable
Disabled by default.
9.
NOTE:
• To enable the QSIG tunneling function, you must also execute the qsig-tunnel enable command
on the voice trunks of both the ingress and egress gateways.
• When the QSIG tunneling function is enabled, the system will not process or send non-QSIG
ISDN calls.
• The overlap mode ISDN calls do not support QSIG tunnel function.
Configuring digital E&M signaling
Configuring a start mode
Similar to analog E&M signaling, digital E&M signaling also provides three start modes:
immediate-start, wink-start and delay-start. The time sequence of digital E&M signaling is the same
as that of analog E&M signaling for all three start modes. The only difference is that analog E&M
81
signaling transmits signaling information through the level change of Tip and Ring line but digital
E&M signaling adopts 4 bits of TS16 to transmit signaling information in the same way as R2
signaling.
To configure the immediate start mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
E&M signaling and select the
immediate start mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
e&m-immediate
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter digital E&M voice
subscriber line view.
subscriber-line
slot-numbe:ts-set-number
N/A
6.
Configure the delay that the
calling party must experience
before sending DTMF digits
in immediate start mode.
delay send-dtmf millseconds
Optional.
The default is 300
milliseconds.
For more information about the timer dial-interval, timer wait-digit, timer ring-back, and delay
send-dtmf commands, see Voice Command Reference.
To configure the delay start mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
E&M signaling and select the
delay start mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
e&m-delay
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter digital E&M voice
subscriber line view.
subscriber-line
slot-numbe:ts-set-number
N/A
6.
Configure the maximum
duration of delay signal in
delay start mode.
delay hold millseconds
Configure the delay that the
called party must wait before
sending a delay signal after it
detects a seizure signal.
delay rising millseconds
7.
Optional.
The default is 400 milliseconds.
Optional.
The default is 300 milliseconds.
To configure the wink start mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
E&M signaling and select the
immediate start mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
e&m-wink
N/A
Exit E1/T1 interface view.
quit
N/A
4.
82
Step
Command
Remarks
5.
Enter digital E&M voice
subscriber line view.
subscriber-line
slot-numbe:ts-set-number
N/A
6.
Configure the maximum time
the calling party waits for
wink signal after sending the
seizure signal in wink start
mode.
delay wink-rising millseconds
Configure the maximum
duration of the wink signal
sent by the called party in
wink start mode.
delay wink-hold millseconds
Configure the minimum
delay the called party must
wait before sending a wink
signal in wink-start mode.
delay send-wink millseconds
7.
8.
Optional.
The default is 2,000 milliseconds.
Optional.
The default is 500 milliseconds.
Optional.
The default is 200 milliseconds.
Enabling E&M non-signaling mode
Analog E&M provides only one private line, which cannot meet the requirement of large-capacity
private lines. Therefore, digital E&M non-signaling should be supported by the VE1/VT1 interface.
The calling process is similar to that of analog E&M signaling except that digital signals are used
instead of analog signals. In this way, a VE1 interface can provide 30 private lines, and a VT1
interface can provide 23 private lines, thereby enlarging the system capacity.
This feature should be used with the PLAR function, you must also execute the private-line
command on the calling voice gateway. For more information about the PLAR function, see
"Configuring dial plans."
Before you enable E&M non-signaling mode, E&M signaling must operate in immediate start mode.
For more information about the analog E&M non-signaling mode, see "Configuring E&M voice
subscriber line."
To enable E&M non-signaling mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter digital E&M voice subscriber
line view.
subscriber-line slot-number:
ts-set-number
N/A
Optional.
3.
Enable E&M non-signaling mode.
open-trunk { caller monitor
interval | called }
Disabled by default.
For more information about the
open-trunk command, see
Voice Command Reference.
Configuring receive and transmit signaling
The ABCD bit pattern of the receive idle signal from the local end must be the same as that of the
transmit idle signal from the remote end. Seized signals and idle signals are processed in the same
way.
To configure receive and transmit signaling:
83
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
E&M signaling and select a
start mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink }
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter digital E&M voice
subscriber line view.
subscriber-line
slot-number:ts-set-number
N/A
6.
Configure the ABCD bit
pattern of receive idle signal.
signal-value received idle ABCD
Configure the ABCD bit
pattern of receive seized
signal.
signal-value received seize ABCD
Configure the ABCD bit
pattern of transmit idle
signal.
signal-value transmit idle ABCD
Configure the ABCD bit
pattern of transmit seized
signal.
signal-value transmit seize ABCD
7.
8.
9.
Optional.
The default is 1101.
Optional.
The default is 0101.
Optional.
The default is 1101.
Optional.
The default is 0101.
IMPORTANT:
After you change the ABCD bit pattern of a digital E&M signal, shut down the digital E&M voice
subscriber line with the shutdown command and then bring the line up with the undo shutdown
command. Otherwise, the voice subscriber line cannot work correctly.
Configuring the time adjustment function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
E&M signaling and select a
start mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ e&m-delay | e&m-immediate |
e&m-wink }
N/A
4.
Exit E1/T1 interface view.
quit
N/A
5.
Enter digital E&M voice
subscriber line view.
subscriber-line
slot-number:ts-set-number
N/A
6.
Configure the maximum
interval between any two
digits of a dialed number.
timer dial-interval seconds
Configure the maximum time
the calling party waits for a
ringback response.
timer ring-back seconds
Configure the maximum time
the called party waits for the
first digit.
timer wait-digit { seconds |
infinity }
7.
8.
84
Optional.
The default is 10 seconds.
Optional.
The default is 60 seconds.
Optional.
The default is 5 seconds.
Querying the circuits of a timeslot or a range of timeslots
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
E&M signaling and select a
start mode.
timeslot-set ts-set-number timeslot-list
timeslots-list signal { e&m-delay |
e&m-immediate | e&m-wink }
N/A
4.
Enter digital E&M signaling
view.
cas ts-set-number
N/A
5.
Query the circuits of a
timeslot or a range of
timeslots.
ts query timeslots timeslots-list
Optional.
Configuring digital LGS signaling
Configuration the time adjustment function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set, enable
digital LGS signaling and set
its start mode.
timeslot-set ts-set-number
timeslot-list timeslots-list signal
{ fxo-ground | fxs-ground |
fxo-loop | fxs-ground |
fxs-loop }
Optional.
4.
Quit digital LGS signaling
view.
quit
N/A
5.
Enter voice subscriber line
view.
subscriber-line:
ts-set-number:ts-set-number
N/A
6.
Configure the dial delay.
delay start-dial value
7.
Configure the maximum time
waiting for the subscriber to
dial the first digit.
timer first-dial seconds
Configure the maximum
interval between any two
dialed digits.
timer dial-interval seconds
Configure the maximum time
that the calling party waits for
a ringback response.
timer ring-back seconds
8.
9.
Optional.
1 second by default.
Optional.
10 seconds by default.
Optional.
10 seconds by default.
Optional.
60 seconds by default.
Querying the circuits of a timeslot or a range of timeslots
Step
Command
Remarks
1.
system-view
N/A
Enter system view.
85
Step
Command
Remarks
2.
Enter E1/T1 interface view.
controller { e1 | t1 } slot-number
N/A
3.
Create a TS set for digital
LGS signaling and select a
start mode.
timeslot-set ts-set-number timeslot-list
timeslots-list signal { fxo-ground | fxo-loop |
fxs-ground | fxs-loop }
N/A
4.
Enter digital signaling view.
cas ts-set-number
N/A
5.
Query the circuits of a
timeslot or a range of
timeslots.
ts query timeslots timeslots-list
Optional.
Configuring a BSV BRI interface
BSV interface
The BRI S/T voice (BSV) interface supports transmission of voice and data, can receive, send,
compress, de-compress digital PCM voice traffic on ISDN BRI interfaces, and realizes VoIP function
through other WAN interfaces of the router.
Generally, a BSV interface is used to connect an ISDN digital telephone, and also can be used as a
trunk interface connecting to a PBX digital trunk. If it cooperates with an FXS or FXO interface, a
BSV interface can realize flexible route selection policies for voice calls.
Configuration prerequisites
The router is equipped with an applicable BSV interface card.
Configuration procedure
To configure a BSV BRI interface:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter the specified BSV BRI
interface view.
interface bri interface-number
N/A
3.
Configure an interface
description.
description text
4.
Enable B channel loopback
detection.
loopback { b1 | b2 | both }
5.
Set the MTU of the BSV BRI
interface.
mtu size
6.
Set the interval for sending
keepalive packets.
timer hold seconds
7.
Enable the interface to
generate linkUp/linkDown
traps upon link changes.
enable snmp trap updown
Optional.
By default, the description of a
voice subscriber line is
interface-name+interface.
Optional.
Not enabled by default.
Optional.
Optional.
86
By default, the polling interval is
10 seconds.
Optional.
Enabled by default.
Step
8.
9.
Command
Remarks
Restore the default
configuration of the BSV BRI
interface.
default
Optional.
Shut down the BSV BRI
interface.
shutdown
Optional.
The BSV BRI interface is enabled
by default.
For more information about the description, loopback, mtu, and shutdown commands, see
Interface Command Reference.
Configuring AMD
Answering machine detection (AMD) is used to detect if a human voice or an answering machine is
answering the call by using a statistic algorithm. AMD is mainly applied in the predictive dialing
system in call centers, as some services need to recognize who is answering the call: a human being,
an answering machine, a fax machine, or a Modem.
AMD uses the answering rhythm (voice duration and silent time) as the statistic algorithm. Generally,
if the speaker is a human voice, there is always a period of silence after a short greeting. If the
speaker is an answering machine, there is always a short silence after a comparatively long greeting.
Enabling the AMD function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line
slot-number:ts-set-number
N/A
Disabled by default.
3.
Enable the AMD function.
amd enable
The AMD detection results are
subject to the language
characteristics, answering
machines, and background
noises and music.
NOTE:
Support for the AMD function depends on the voice card.
Configuring AMD parameters
The accuracy of AMD detection results depends on the settings of detection parameters. You can
increase the accuracy of the detection by adjusting AMD parameters.
To configure AMD parameters:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure AMD parameters.
amd parameter { machine-time
value | max-analyze-time value |
Optional.
87
By default, the machine-time
Step
Command
Remarks
min-silence-time value |
valid-voice-time value |
voice-energy-threshold value }
keyword is 2600 milliseconds, the
max-analyze-time keyword is
4000 milliseconds, the
min-silence-time keyword is 800
milliseconds, the
valid-voice-time keyword is 120
milliseconds, the
voice-energy-threshold
keyword is 100.
For more information about the
parameters, see Voice Command
Reference.
NOTE:
The execution of the amd parameter command in voice view is a global configuration, which means
it will be effective for the entire device.
Configuring reverse charging function
Typically, the calling party is charged to place a call to the called party. If you want to charge the
called party, you can configure a prefix to match the called number for reverse charging.
After you configure this function, the router identifies a reverse-charge call by comparing the called
number of an incoming call with a prefix. If the called number matches the prefix, the router interacts
with the PBX device to charge the called party.
If you configure the reverse charging function in both voice view and digital voice subscriber line view,
the configuration in digital voice subscriber line view takes effect.
To configure the reverse charging function in voice view:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure a prefix for reverse
charging.
reverse-charge prefix string
By default, no prefix for reverse
charging is configured.
To configure the reverse charging function in digital voice subscriber line view:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter digital voice subscriber
line view.
subscriber-line line-number
N/A
3.
Configure a prefix for reverse
charging.
reverse-charge prefix string
By default, no prefix for reverse
charging is configured.
NOTE:
Support for the reverse charging function depends on the voice card.
88
Enabling the router to treat DISCONNECT
messages with PI 8 as standard DISCONNECT
messages
When the router receives a standard DISCONNECT message, it brings down the user-side
B-channel, sends a RELEASE message, and disconnects the call. When it receives a
DISCONNECT message with Progress Indicator (PI) 8, it brings up the B-channel to receive inband
information. The router does not disconnect the call or release the B-channel until it receives a
standard DISCONNECT message.
To avoid unnecessarily occupying B-channel resources, enable the router to treat DISCONNECT
messages with PI 8 as standard DISCONNECT messages.
To enable the router to treat DISCONNECT messages with PI 8 as standard DISCONNECT
messages:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enable the router to treat
DISCONNECT messages
with PI 8 as standard
DISCONNECT messages.
call disc-pi-off
By default, the router does not
disconnect a call when it receives
a DISCONNECT message with PI
8.
Mirroring PCM, RTP packets, or voice command
data on a digital voice subscriber line
The mirroring function copies the specified PCM, RTP, and voice command data on a digital voice
subscriber line to the specified destination. With the mirroring function, you can analyze and locate
problems.
To mirror PCM, RTP, or voice command data on a digital voice subscriber line to a specified interface
or destination:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter digital voice
subscriber line view.
subscriber-line line-number
N/A
•
3.
Mirror PCM, RTP, or
voice command data to
a specified interface or
destination.
•
•
Mirror PCM or RTP data based on a calling
number:
mirror number number { pcm | { in | out |
all } data } calling calling-number to
{ local-interface interface-type
interface-number [ mac H-H-H ] | remote-ip
ip-address [ port port ] }
Mirror PCM data based on a timeslot:
mirror number number pcm { bdsp | fdsp }
channel-number to { local-interface
interface-type interface-number [ mac
H-H-H ] | remote-ip ip-address [ port port ] }
Mirror RTP or voice command data on all
89
Optional.
By default, no traffic is
mirrored.
Step
Command
Remarks
timeslots:
mirror number number { in | out | all }
{ command | data } to { local-interface
interface-type interface-number [ mac
H-H-H ] | remote-ip ip-address [ port port ] }
Displaying and maintaining digital voice
subscriber lines
Task
Command
Remarks
Display configurations of digital
voice subscriber lines.
display voice subscriber line
[ slot-number:{ { ts-set-number |
ts-set-number.sub-timeslot } | 15 | 23 } ]
[ brief ] [ | { begin | exclude | include }
regular-expression ]
Available in any view.
Digital voice subscriber line configuration
examples
This section provides configuration examples for digital voice subscriber lines.
E1 R2 signaling and digital E&M signaling configuration
example
Network requirements
As shown in Figure 23, Telephones in City A and City B communicate with each other through voice
routers (Router A and Router B) across an IP network, as shown in the network diagram.
•
In City A, Router A is connected to a PBX with an E1 subscriber line on which R2 signaling
travels, and to the telephone at 0101003 with an FXS voice subscriber line.
•
In City B, Router B is connected to a PBX with an E1 subscriber line on which digital E&M
signaling (in the delay start mode) travels.
One-stage dialing mode is configured on the two routers.
Figure 23 Network diagram
90
Configuration procedure
1.
Configure Router A:
# Configure the IP address 1.1.1.1/24 for the interface Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 1.1.1.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Create a TS set on the interface Ethernet 1/1.
<RouterA> system-view
[RouterA] controller e1 1/1
[RouterA-E1 1/1] timeslot-set 1 timeslot-list 1-31 signal r2
[RouterA-E1 1/1] quit
# Create a POTS voice entity corresponding to telephone number 010-1003 for the FXS
interface.
[RouterA] system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1003 pots
# Configure a target match-template for the POTS voice entity.
[RouterA-voice-dial-entity1003] match-template 0101003
# Associate the POTS voice entity with FXS subscriber line 3/0.
[RouterA-voice-dial-entity1003] line 3/0
[RouterA-voice-dial-entity1003] quit
# Create a POTS voice entity for the E1 interface.
[RouterA-voice-dial] entity 1001 pots
# Configure a target match-template pointing to telephone number 010-1001 for the POTS
voice entity.
[RouterA-voice-dial-entity1001] match-template 0101001
# Associate the POTS voice entity with subscriber line 1/1:1.
[RouterA-voice-dial-entity1001] line 1/1:1
[RouterA-voice-dial-entity1001] send-number all
[RouterA-voice-dial-entity1001] quit
# Create a POTS voice entity corresponding to telephone number 010-1002 for the E1
interface.
[RouterA-voice-dial] entity 1002 pots
# Configure a target match-template for the POTS voice entity.
[RouterA-voice-dial-entity1002] match-template 0101002
# Associate the POTS voice entity with subscriber line 1/1:1.
[RouterA-voice-dial-entity1002] line 1/1:1
[RouterA-voice-dial-entity1002] send-number all
# Create a VoIP voice entity.
[RouterA-voice-dial-entity1002] entity 0755 voip
# Configure a target match-template for the VoIP voice entity.
[RouterA-voice-dial-entity755] match-template 0755....
# Configure the target address of the VoIP voice entity.
[RouterA-voice-dial-entity755] address ip 2.2.2.2
2.
Configure Router B:
91
# Configure the IP address 2.2.2.2/24 for the interface Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 2.2.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Create a TS set on interface E1 1/1.
[RouterB] system-view
[RouterB] controller e1 1/1
[RouterB-E1 1/1] timeslot-set 1 timeslot-list 1-31 signal e&m-delay
[RouterB-E1 1/1] quit
# Create a POTS voice entity corresponding to telephone number 0755-2001 for the E1
interface.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2001 pots
# Configure a target match-template for the POTS voice entity.
[RouterB-voice-dial-entity2001] match-template 07552001
# Associate the POTS voice entity with FXS subscriber line 1/1:1.
[RouterB-voice-dial-entity2001] line 1/1:1
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] quit
# Create a POTS voice entity corresponding to telephone number 0755-2002 for the E1
interface.
[RouterB-voice-dial] entity 2002 pots
# Configure a target match-template for the POTS voice entity.
[RouterB-voice-dial-entity2002] match-template 07552002
# Associate the POTS voice entity with subscriber line 1/1:1.
[RouterB-voice-dial-entity2002] line 1/1:1
[RouterB-voice-dial-entity2002] send-number all
[RouterB-voice-dial-entity2002] quit
# Create a VoIP voice entity.
[RouterB-voice-dial] entity 010 voip
# Configure a target match-template for the VoIP voice entity.
[RouterB-voice-dial-entity10] match-template 010....
# Configure the target address of the VoIP voice entity.
[RouterB-voice-dial-entity10] address ip 1.1.1.1
E1 voice DSS1 signaling configuration example
Network requirements
As shown in Figure 24, Telephones in City A and City B communicate with each other through voice
routers (Router A and Router B) across an IP network.
•
In City A, Router is connected to a PBX through an E1 subscriber line, and to the telephone at
0101003 through an FXS voice subscriber line.
•
In City B, Router B is connected only to a PBX through an E1 subscriber line.
The two routers communicate with their respective PBX by exchanging DSS1 user signaling through
an ISDN interface. One-stage dialing mode is configured on the two routers.
92
Figure 24 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure the IP address 1.1.1.1/24 for the interface Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 1.1.1.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Create an ISDN PRI group on interface E1 1/1.
[RouterA] system-view
[RouterA] controller e1 1/1
[RouterA-E1 1/1] pri-set
# Create a POTS voice entity for the FXS interface.
[RouterA] system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1003 pots
# Configure a target match-template pointing to telephone number 010-1003 for the POTS
voice entity.
[RouterA-voice-dial-entity1003] match-template 0101003
# Associate the POTS voice entity with FXS subscriber line 3/0.
[RouterA-voice-dial-entity1003] line 3/0
[RouterA-voice-dial-entity1003] quit
# Create a POTS voice entity for the ISDN PRI interface.
[RouterA-voice-dial] entity 1001 pots
# Configure a target match-template pointing to telephone number 010-1001 for the POTS
voice entity.
[RouterA-voice-dial-entity1001] match-template 0101001
# Associate the POTS voice entity with subscriber line 1/1:15.
[RouterA-voice-dial-entity1001] line 1/1:15
[RouterA-voice-dial-entity1001] send-number all
[RouterA-voice-dial-entity1001] quit
# Create a POTS voice entity for the ISDN PRI interface.
[RouterA-voice-dial] entity 1002 pots
# Configure a target match-template pointing to telephone number 010-1002 for the POTS
voice entity.
[RouterA-voice-dial-entity1002] match-template 0101002
93
# Associate the POTS voice entity with subscriber line 1/1:15.
[RouterA-voice-dial-entity1002] line 1/1:15
[RouterA-voice-dial-entity1002] send-number all
[RouterA-voice-dial-entity1002] quit
# Create a VoIP voice entity.
[RouterA-voice-dial] entity 0755 voip
# Configure a target match-template for the VoIP voice entity.
[RouterA-voice-dial-entity755] match-template 0755....
# Configure the target address of the VoIP voice entity.
[RouterA-voice-dial-entity755] address ip 2.2.2.2
2.
Configure Router B:
# Configure the IP address 2.2.2.2/24 for the interface Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 2.2.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Create an ISDN PRI group on interface E1 1/1.
[RouterB] system-view
[RouterB] controller e1 1/1
[RouterB-E1 1/1] pri-set
[RouterB-E1 1/1] quit
# Create a POTS voice entity for the ISDN PRI interface.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2001 pots
# Configure a target match-template pointing to telephone number 0755-2001 for the POTS
voice entity.
[RouterB-voice-dial-entity2001] match-template 07552001
# Associate the POTS voice entity with subscriber line 1/1:15.
[RouterB-voice-dial-entity2001] line 1/1:15
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] quit
# Create a POTS voice entity for the ISDN PRI interface.
[RouterB-voice-dial] entity 2002 pots
# Configure a target match-template pointing to telephone number 0755-2002 for the POTS
voice entity.
[RouterB-voice-dial-entity2002] match-template 07552002
# Associate the POTS voice entity with subscriber line 1/1:15.
[RouterB-voice-dial-entity2002] line 1/1:15
[RouterB-voice-dial-entity2002] send-number all
[RouterB-voice-dial-entity2002] quit
# Create a VoIP voice entity.
[RouterB-voice-dial] entity 010 voip
# Configure a target match-template for the POTS voice entity.
[RouterB-voice-dial-entity10] match-template 010....
# Configure the target address of the VoIP voice entity.
[RouterB-voice-dial-entity10] address ip 1.1.1.1
94
QSIG tunneling configuration example
Network requirements
As shown in Figure 25, connect Router A and Router B each to a PBX through QSIG signaling. The
SIP protocol runs between Router A and Router B.
The requirements are as follows:
When a call is placed from 010-1001 to 0755-2001, the ingress gateway Router A encapsulates the
received QSIG signaling messages into SIP messages, and sends the SIP messages to the egress
gateway Router B over the SIP network. This extracts the QSIG signaling messages from the SIP
messages and sends them to the receiving end at the ISDN side.
Figure 25 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure IP address 1.1.1.1/24 for interface Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 1.1.1.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Create an ISDN PRI group on interface E1 1/1.
[RouterA] system-view
[RouterA] controller e1 1/1
[RouterA-E1 1/1] pri-set
[RouterA-E1 1/1] quit
# Set the ISDN protocol type and protocol mode for the ISDN interface serial 1/1:15.
[RouterA] interface serial 1/1:15
[RouterA-Serial1/1:15] isdn protocol-type qsig
[RouterA-Serial1/1:15] quit
# Create a POTS voice entity for the ISDN PRI interface.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1001 pots
# Configure a target match-template pointing to telephone number 010-1001 for the POTS
voice entity.
[RouterA-voice-dial-entity1001] match-template 0101001
# Associate the POTS voice entity with subscriber line 1/1:15.
[RouterA-voice-dial-entity1001] line 1/1:15
[RouterA-voice-dial-entity1001] send-number all
95
[RouterA-voice-dial-entity1001] quit
[RouterA-voice-dial] quit
[RouterA-voice] quit
# Enable the QSIG tunneling function on the subscriber line associated with the POTS voice
entity.
[RouterA] subscriber-line 1/1:15
[RouterA-subscriber-line1/1:15] qsig-tunnel enable
[RouterA-subscriber-line1/1:15] quit
# Create a VoIP voice entity.
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 0755 voip
# Configure a target match-template for the VoIP voice entity.
[RouterA-voice-dial-entity755] match-template 0755....
# Configure the target address of the VoIP voice entity.
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
2.
Configure Router B:
# Configure the IP address 2.2.2.2/24 for the interface Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 2.2.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Create an ISDN PRI group on interface E1 1/1.
[RouterB] controller e1 1/1
[RouterB-E1 1/1] pri-set
[RouterB-E1 1/1] quit
# Set the ISDN protocol type and protocol mode for the ISDN interface serial 1/1:15.
[RouterB] interface serial 1/1:15
[RouterB-Serial1/1:15] isdn protocol-type qsig
[RouterB-Serial1/1:15] quit
# Create a POTS voice entity for the ISDN PRI interface.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2001 pots
# Configure a target match-template pointing to telephone number 0755-2001 for the POTS
voice entity.
[RouterB-voice-dial-entity2001] match-template 07552001
# Associate the POTS voice entity with subscriber line 1/1:15.
[RouterB-voice-dial-entity2001] line 1/1:15
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] quit
[RouterB-voice-dial] quit
# Enable the QSIG tunneling function on the subscriber line associated with the POTS voice
entity.
[RouterB] subscriber-line 1/1:15
[RouterB-subscriber-line1/1:15] qsig-tunnel enable
[RouterB-subscriber-line1/1:15] quit
# Create a VoIP voice entity.
[RouterB-voice] dial-program
96
[RouterB-voice-dial] entity 010 voip
# Configure a target match-template for the VoIP voice entity.
[RouterB-voice-dial-entity10] match-template 010....
# Configure the target address of the VoIP voice entity.
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
Troubleshooting digital voice subscriber line
configuration
Failure of call connection from router to PSTN
Symptom
With R2 signaling adopted, the router cannot establish connection with the subscriber at the switch
side.
Solution
Use the display current-configuration command to check that the trunking mode on the router
matches that on the switch. When the switch adopts outgoing trunking mode, the router must adopt
incoming or bidirectional trunking mode. When the switch adopts incoming trunking mode, the router
must adopt outgoing or bidirectional trunking mode. When the router adopts incoming trunking mode,
it only accepts incoming calls.
97
Configuring dial plans
Overview
More requirements on dial plans arise with the wide application of VoIP. A desired dial plan should be
flexible, reasonable and operable, and be able to help a voice gateway manage numbers in a unified
way, making number management more convenient and reasonable.
The dial plan process on the calling side differs from that on the called side. The following discusses
these two dial plan processes.
Dial plan process
On the calling side
Figure 26 shows the dial plan operation process on the calling side.
Figure 26 Flow chart for dial plan operation process on the calling side
1.
The voice gateway on the calling side replaces the calling and called numbers according to the
number substitution rule on the receiving line.
2.
The voice gateway performs global number substitution.
3.
The gateway selects a proper voice entity based on the voice entity selection priority rules and
replaces the calling and called numbers.
4.
The gateway initiates a call to the called side and sends the calling and called numbers.
98
On the called side
Figure 27 shows the dial plan operation process on the called side.
Figure 27 Flow chart for dial plan operation process on the called side
1.
After receiving a voice call (the called number), the voice gateway on the called side performs
global calling/called number substitution.
2.
The voice gateway on the called side selects proper voice entities based on the voice entity
selection priority rules. (Number substitution might also be involved during the voice entity
selection.) If the called party is a local voice subscriber, the gateway directly connects the
subscriber line. If the called party is a PSTN subscriber, the gateway initiates a call and sends
the calling and called numbers to the PSTN. The PBX in the PSTN connects the call.
Regular expression
You will frequently use some regular expressions when you configure number substitution rules.
Regular expressions are a powerful and flexible tool for pattern matching and substitution. They are
not restricted to a language or system and are widely accepted.
When using a regular expression, you need to construct a matching pattern according to certain
rules, and then compare the matching pattern with the target object. The simplest regular
expressions do not contain any metacharacter. For example, you can specify the regular expression
"hello", which matches only the string "hello".
To help you construct flexible matching patterns, regular expressions support some special
characters, called metacharacters, which define the way other characters appear in the target
object. Table 12 describes these metacharacters.
Table 12 Metacharacters
Metacharacter
Meaning
0-9
Digits 0 through 9.
Pound sign (#) and
asterisk (*)
Each indicates a valid digit.
99
Metacharacter
Meaning
Dot (.)
Wildcard, which can match any valid digit. For example, 555…. can match any
number beginning with 555 and ending in four additional characters.
Hyphen (-)
Used to connect two numbers (The smaller comes before the larger) to indicate a
range of numbers, for example, 1-9 inclusive.
Brackets ([ ])
Delimits a range for matching. It can be used together with signs such as !, %, and +.
For example, [235-9] indicates one number of 2, 3, and 5 through 9.
Parentheses (( ))
Indicates a sub-expression. For example, (086) indicates the character string 086. It
is usually used together with signs such as !, %, and +. For example, (086)!010 can
match two character strings 010 and 086010.
Exclamation point (!)
A control character, indicating that the sub-expression before it appears once or
does not appear. For example, (010)!12345678 can match 12345678 and
01012345678.
Plus sign (+)
A control character, indicating that the sub-expression before it appears one or more
times. However, if a calling number starts with the plus sign, the sign itself does not
have special meanings and only indicates that the following is an effective number
and the whole number is E.164-compliant. For example, 9876(54)+ can match
987654, 98765454, 9876545454, and so on, and +110022 is an E.164-compliant
number.
Percent sign (%)
A control character, indicating that the sub-expression before it appears multiple
times or does not appear. For example, 9876(54)% can match 9876, 987654,
98765454, 9876545454, and so on.
The sub-expression (one digit or digit string) before a control character such as exclamation point (!),
plus sign (+), and percent sign (%) can appear for the corresponding times indicated by the control
character. For example, (100)+ can match 100, 100100, 100100100, and so on. Once any number of
them is matched, the match is considered an exact match. In the longest match mode, the voice
gateway will ignore subsequent digits dialed by the subscriber after an exact match. (For the case
that the gateway needs to wait for subscribers to continue dialing after an exact match, see the
description for T mode.)
The characters (\) and (|) are mainly used in regular expressions and cannot be used as common
characters. The character (\) is an escape character. If you want a control character to represent
itself, you need to add the escape character (\) before it. For example, (\+) represents the character
(+) itself because (+) is a control character in regular expressions. The character (|) means that the
current character (string) is the character (string) on either the left or the right. For example,
0860108888|T means that the current character string is either 0860108888 or T.
T mode—The character T in the match-template match-string means that the voice gateway should
wait for more digits until the number exceeds the maximum length or the dial timer expires.
If a number starts with the plus sign (+), note the following when you use it on a trunk: the E&M, R2,
and LGS signaling uses DTMF, and as the plus sign (+) does not have a corresponding audio, the
number cannot be transmitted to the called side successfully. While the DSS1 signaling uses ISDN
transmission, the above problem does not exist. Therefore, you should avoid using a number that
cannot be identified by the signaling itself; otherwise, the call will fail.
Introduction to number substitution
According to the network requirements, you can first configure a number substitution rule list, and
then define specific number substitution rules, dot-match rules, and preferred number substitution
rules for the list. Finally, you can apply these substitution rules globally or to voice entities and voice
subscriber lines to substitute calling/called numbers flexibly.
If there exist multiple number substitution rules in a number substitution rule list, only one number
substitution rule will be matched. The match process is as follows:
100
1.
The preferred number substitution rule is matched first. If the match succeeds, the gateway
substitutes numbers based on this rule.
2.
If the match fails, the gateway matches other number substitution rules in sequence. Once a
rule is matched successfully, the gateway stops matching other number substitution rules.
Here, the dot represents virtually matched digits. Virtually matched digits, including dot (.), plus sign
(+), percent sign (%), exclamation points (!), and brackets ([]), see those that match the variable part
in a regular expression. For example, the virtually matched digits are the digit "2", the digit "5", and
the digits "25" respectively when the number 1255 matches the regular expressions 1[234]55, 125+,
and 1..5.
Hardware compatibility with dial plans
Dial plans are not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
Configuration task list
Task
Remarks
Configuring calling numbers permitted to call out
Optional.
Configuring call authority control
Optional.
Configuring a number match mode
Optional.
Configuring match order of voice entity selection rules
Optional.
Configuring a number priority peer
Optional.
Configuring a maximum-call-connection set
Optional.
Configuring number substitution
Optional.
Configuring a number sending mode
Optional.
Configuring a dial prefix
Optional.
Configuring calling numbers permitted to call out
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
dial-program
N/A
101
Step
Command
Remarks
entity entity-number { pots | ivr |
vofr | voip }
N/A
view.
4.
Create a voice entity and
enter voice entity view.
5.
Configure calling numbers
permitted to call out.
Optional.
caller-permit calling-string
By default, no calling number is
configured, and outgoing calls are
not restricted.
NOTE:
The calling-string argument is in the format of { [ + ] string [ $ ] }| $. For specific meanings of these
symbols in the format, see Voice Command Reference.
Configuring call authority control
To configure call authority control, you can assign subscribers to a subscriber group, then bind the
group, which has permit or deny authority configured, to a voice entity.
When a subscriber originates a call that matches the voice entity, the system compares the calling
number with each number in the bound subscriber group. If a match is found, the calling is permitted
or denied according to the authority (permit or deny) configured; otherwise, the system finds the
next matching voice entity until the call is permitted or denied.
Configuring match templates for a subscriber group
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter subscriber group view.
subscriber-group list-number
At most ten subscriber groups
can be configured for the system.
5.
Configure a subscriber group
description string.
description text
6.
Configure match templates for
the subscriber group.
Optional.
By default, no subscriber group
description string is configured.
Optional.
match-template match-string
By default, no match template is
configured for the subscriber
group.
Binding a subscriber group to a voice entity
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Create a voice entity and enter
entity entity-number { pots | ivr
N/A
102
Step
5.
Command
voice entity view.
| vofr | voip }
Bind a subscriber group to the
voice entity.
caller-group { deny | permit }
subscriber-group-list-number
Remarks
By default, no subscriber group is
bound to the voice entity, that is,
any calling number is allowed to
originate calls.
Enabling private line auto ring-down
With the private line auto ring-down (PLAR) function enabled, the voice gateway automatically dials
the specified called number (string) as soon as the subscriber picks up the phone.
To configure the PLAR function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable the PLAR function.
private-line string
Optional.
Disabled by default.
Configuring a number match mode
This section describes the procedures for configuring a global number match mode and a dial
terminator.
Configuration prerequisites
The required basic configurations have been completed on POTS, VoIP, VoFR, and (IVR entities.
Configuring a global number match mode
Use the number-match command to determine the number match mode: longest match or shortest
match. Suppose you have configured match-template 0106688 and match-template 01066880011
respectively on two voice entities.
When a subscriber dials 01066880011:
•
If the router is configured to use the shortest match mode, the dialed number will match
0106688. That is, the router will establish a call connection to 0106688 at the remote end,
without processing the last four digits 0011.
•
If the router is configured to use the longest match mode, the dialed number will match
01066880011. That is, the router will establish a call connection to 01066880011 at the remote
end.
When a subscriber dials 0106688:
•
If the router is configured to use shortest match mode, it will match match-template 0106688.
•
If the router is configured to use longest match mode, it will wait for further digits. After the dial
timer expires, the router will ignore the configured longest match mode and automatically use
shortest match to establish a call connection.
103
When a subscriber dials 0106688#, if you configure the router to use longest match mode and
the dial terminator "#" on the router, the router will as well ignore the configured longest match
mode and use shortest match mode to establish a call connection.
To configure a global number match mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Configure a global number
match mode.
number-match { longest |
shortest }
By default, the shortest match
mode is adopted.
Configuring a dial terminator
In areas where variable-length numbers are used, you can specify a character as the dial terminator
so that the voice gateway can dial out the number before the dialing interval expires. The dial
terminator identifies the end of a dialing process, and a call connection will be established based on
the received digits when the dial terminator is received. The voice gateway will not wait for further
digits even if the longest match mode has been globally configured.
To configure a dial terminator:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
Optional.
By default, no dial terminator is
configured.
4.
Configure a dial terminator.
terminator character
If you set the argument character
to # or *, and if the first character
of the configured entity number is
the same as the argument
character (# or *), the device will
take this first character as a
common number rather than a dial
terminator.
Number match mode configuration example
Figure 28 and the following describe the configurations for different number match modes on Router
A and Router B:
104
Figure 28 Network diagram
•
Shortest number match
a. Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
# Configure POTS entity 1000.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 10001234$
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] quit
# Configure VoIP entity 2000 and VoIP entity 2001.
[RouterA-voice-dial] entity2000 voip
[RouterA-voice-dial-entity2000] match-template 20001234$
[RouterA-voice-dial-entity2000] address ip 1.1.1.2
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity2001 voip
[RouterA-voice-dial-entity2001] match-template 200012341234$
[RouterA-voice-dial-entity2001] address ip 1.1.1.2
[RouterA-voice-dial-entity2001] quit
b. Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
# Configure POTS entity 2000 and POTS entity 2001.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 20001234$
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] quit
[RouterB-voice-dial] entity2001 pots
[RouterB-voice-dial-entity2001] match-template 200012341234$
[RouterB-voice-dial-entity2001] line 1/1
After you dial number 20001234 at Telephone A, the number matches VoIP entity 2000 and
Telephone B is alerted because the device adopts the shortest match mode by default.
•
Longest number match
# Configure the longest match mode on Router A. The other steps are the same as those for the
shortest match mode.
105
[RouterA-voice-dial] number-match longest
After you dial number 20001234 at Telephone A and waits for a period of time (during this
period, you can continue dialing), the number matches VoIP entity 2000 and Telephone B is
alerted. If you continue to dial 1234 during this period (that is, the dialed number is actually
200012341234), the number matches VoIP entity 2001 and Telephone C is alerted.
•
Dial terminator
# Configure the longest match mode and the dial terminator # on Router A. The other steps are
the same as those for the shortest match mode.
[RouterA-voice-dial] number-match longest
[RouterA-voice-dial] terminator #
After you dial 20001234# at Telephone A, the number immediately matches VoIP entity 2000
and Telephone B is alerted.
Configuring match order of voice entity selection
rules
If multiple voice entities can match a call number, the voice gateway follows the configured rules to
select a proper voice entity.
The match order is as follows:
1.
Voice entity type selection priority rule—This rule means that different priorities are
configured for different types of voice entities (VoIP, POTS, VoFR and IVR). The voice gateway
matches a voice entity according to the priorities of different types of voice entities.
2.
Match order of voice entity selection rules—These rules cover exact match, priority, random
selection, and longest idle time. For more information, see Voice Command Reference. You
can select one to three rules to form a sequence. The voice gateway will first select a voice
entity according to the first rule. If the voice gateway fails to decide which number should be
selected according to the first rule, it will apply the second rule, and so on.
Configuration prerequisites
The required basic configurations have been completed on POTS, VoIP, VoFR, and IVR entities.
Configuration procedure
To configure voice entity selection priority rules:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots | ivr |
vofr | voip }
N/A
5.
Configure the voice entity
priority.
priority priority-order
6.
Exit voice entity view.
quit
N/A
7.
Configure match order of
voice entity selection rules.
select-rule rule-order 1st-rule
Optional.
106
Optional.
0 by default.
Step
Command
Remarks
[ 2nd-rule [ 3rd-rule ] ]
By default, the match order of
rules for the voice entity selection
is exact match->voice entity
priority->random selection.
Configuration example of voice entity selection priority rules
Figure 29 and the following describe the voice entity selection priority rule configuration on Router A
and Router B.
Figure 29 Network diagram
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
# Configure POTS entity 1000.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 10001234$
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] quit
# Configure VoIP entity 2000, VoIP entity 2001, and VoIP entity 2002.
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 20001234$
[RouterA-voice-dial-entity2000] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2000] priority 10
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity2001 voip
[RouterA-voice-dial-entity2001] match-template 2000123.$
[RouterA-voice-dial-entity2001] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2000] priority 5
[RouterA-voice-dial-entity2001] quit
[RouterA-voice-dial] entity2002 voip
[RouterA-voice-dial-entity2002] match-template 2000....$
[RouterA-voice-dial-entity2002] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2002] quit
2.
Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
# Configure POTS entity 2000.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 20001234$
[RouterB-voice-dial-entity2000] line 1/0
107
3.
Configure different voice entity selection priority rules:
{
Configure voice entities to be selected in sequence of exact match, priority, and random
selection.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule rule-order 1 2 3
After Telephone dials 20001234, the number will match VoIP entity 2000.
{
Configure voice entities to be selected in sequence of priority, exact match, and random
selection.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule rule-order 2 1 3
After Telephone A dials 20001234, the number will match VoIP entity 2002.
{
Configure voice entities to be selected at random.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule rule-order 3
After Telephone dials 20001234, the number will match VoIP entity 2000, 2001, or 2002 at
random.
Configuring voice entity type selection priority rules
The voice entity type selection priority rules take precedence over the voice entity selection priority
rules.
To configure voice entity type selection priority rules:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
Optional.
4.
Configure voice entity type
selection priority rules.
select-rule type-first 1st-type
2nd-type 3rd-type [ 4th-type ]
By default, voice entities are not
selected by their types.
The priority of a value with a T (for
example, 3.T) is higher than that
of the same value without a T (for
example, 3).
Configuration example of voice entity type selection priority
rules
There are an IP connection and a PRI connection between Router A and Router B. Figure 30 and the
following describe the configurations for different voice entity type selection priority rules on Router A
and Router B.
108
Figure 30 Network diagram
1.
Configure Router A:
# Configure PRI signaling.
<RouterA> system-view
[RouterA] controller E1 5/0
[RouterA-E1 5/0] pri-set
[RouterA-E1 5/0] quit
[RouterA] interface Serial 5/0:15
[RouterA-Serial5/0:15] isdn protocol-mode network
[RouterA-E1 5/0] quit
[RouterA] voice-setup
[RouterA-voice] dial-program
# Configure POTS entity 1000 and POTS entity 1001.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 10001234$
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] entity1001 pots
[RouterA-voice-dial-entity1001] match-template 20001234$
[RouterA-voice-dial-entity1001] line 5/0:15
[RouterA-voice-dial-entity1001] send-number all
[RouterA-voice-dial-entity1001] quit
# Configure VoIP entity 2000.
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 20001234$
[RouterA-voice-dial-entity2000] address sip ip 1.1.1.2
2.
Configure Router B:
# Configure PRI signaling.
<RouterB> system-view
[RouterB] controller E1 5/0
[RouterB-E1 5/0] pri-set
[RouterB-E1 5/0] quit
[RouterB] voice-setup
[RouterB-voice] dial-program
# Configure POTS entity 2000.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 20001234$
[RouterB-voice-dial-entity2000] line 1/0
3.
Configure different voice entity type selection priority rules:
{
Configure the system to select voice entities in order of VoIP->POTS->VoFR->IVR.
<RouterA> system-view
[RouterA] voice-setup
109
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule type-first
2 1 3 4
After Telephone A dials 20001234, the number will match VoIP entity 2000.
{
Configure the system to select voice entities in order of POTS->VoIP->VoFR->IVR.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule type-first
1 2 3 4
After Telephone A dials 20001234, the number will match POTS entity 1001.
Configuring the voice entity search function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Configure the maximum
number of voice entities
found before a search
process stops.
select-rule search-stop
max-number
5.
Enter voice entity view.
entity entity-number { pots | ivr |
vofr | voip }
6.
Disable the voice entity
search function.
select-stop
Optional.
By default, the maximum number
of voice entities found before a
search process stops is 128.
N/A
Optional.
By default, the voice entity search
function is enabled.
Configuration example of the voice entity search function
Figure 31 and the following describe the configurations for the voice entity search function on Router
A and Router B.
Figure 31 Network diagram
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
# Configure POTS entity 1000.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 10001234$
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] quit
110
# Configure VoIP entities 2000, 2001, and 2002.
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 20001234$
[RouterA-voice-dial-entity2000] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity2001 voip
[RouterA-voice-dial-entity2001] match-template 2000123.$
[RouterA-voice-dial-entity2001] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2001] quit
[RouterA-voice-dial] entity 2002 voip
[RouterA-voice-dial-entity2002] match-template T
[RouterA-voice-dial-entity2002] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2002] quit
2.
Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
# Configure POTS entity 2000.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 20001234$
[RouterB-voice-dial-entity2000] line 1/0
3.
Set the maximum number of voice entities found before a search process stops to 2.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] select-rule search-stop 2
After Telephone dials 20001234, the number will match VoIP entity 2000 by default. You can use the
display voice call-info verbose command to view the other voice entities that meet the matching
conditions.
[RouterA-voice-dial] display voice call-info verbose
The information table for current calls in detail
#
**************** CALL 0 ***************
Call direction
: From CS
ViIfIndex
: 0x002C0060
Related module ==>
Module ID
: LGS
Reference Numbers : 1
Module ID
: CMC
Reference Numbers : 1
Current used voice entity : 2000
Voice entities are offered :
2000
2001
#
End
4.
Restore the maximum number of voice entities found before a search process stops to the
default.
111
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] undo select-rule search-stop
[RouterA-voice-dial] display voice call-info verbose
The information table for current calls in detail
#
**************** CALL 0 ***************
Call direction
: From CS
ViIfIndex
: 0x002C0060
Related module ==>
Module ID
: LGS
Reference Numbers : 1
Module ID
: CMC
Reference Numbers : 1
Current used voice entity : 2000
Voice entities are offered :
2000
2001
2002
#
End
5.
Disable the voice entity search function.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] select-stop
[RouterA-voice-dial-entity2000] display voice call-info verbose
The information table for current calls in detail
#
**************** CALL 0 ***************
Call direction
: From CS
ViIfIndex
: 0x002C0060
Related module ==>
Module ID
: LGS
Reference Numbers : 1
Module ID
: CMC
Reference Numbers : 1
Current used voice entity : 2000
Voice entities are offered :
2000
#
End
112
Configuring a number priority peer
Configuration prerequisites
The required basic configurations have been completed on POTS, VoIP, VoFR, and IVR entities.
Configuration procedure
To configure a number priority peer:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
Optional.
4.
Configure a number priority
peer.
number-priority peer enable
By default, a number starting with
an asterisk (*) or a pound sign (#)
will first match against a service
feature code.
After the number-priority peer enable command is configured, a dialed number will match first
against a voice entity match template and then a service feature code. For example, if a service
feature code is *40*1234 and the match template *40 is configured for a voice entity, *40*1234 dialed
by a user will first match the number template *40 (*40 is dialed out as the called number), and the
feature corresponding to the service feature code *40*1234 will not be triggered.
For more information about the configuration of feature codes, see "Configuring call services."
Configuring a maximum-call-connection set
To control communication traffic, limit the total call connections for one or more voice entities
according to the network scale.
To configure a maximum-call-connection set:
1.
Create a maximum-call-connection set. The parameters include a set label and the maximum
number of call connections.
2.
Bind the maximum-call-connection set to voice entities.
By comparing the maximum number of call connections with the number of existing call connections
of voice entities in a set, the voice gateway determines whether these voice entities can establish
new calls.
Configuration prerequisites
The required basic configurations have been completed on POTS, VoIP, VoFR, and IVR entities.
Configuration procedure
To configure a maximum-call-connection set:
113
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Configure a
maximum-call-connection
set.
max-call set-number
max-number
By default, no
maximum-call-connection set is
configured.
5.
Enter voice entity view.
entity entity-number { pots | ivr |
vofr | voip }
N/A
6.
Bind the
maximum-call-connection
set to the current voice entity.
max-call set-number
By default, no
maximum-call-connection set is
bound to a voice entity.
Configuring number substitution
A number substitution rule list defines some number substitution methods, and is used wherever
number substitution is necessary. There is no limitation on where and how many times it is used.
Therefore, a number substitution rule list might be bound globally and bound to different voice
entities and subscriber lines.
The characteristics of global calling/called number substitution or calling/called number substitution
on voice entities and subscriber lines are as follows:
•
Global number substitution—The voice gateway substitutes calling and called numbers of all
incoming and outgoing calls according to the number substitution rules configured in dial
program view. Multiple number substitution rule lists can be bound for global calling and called
numbers substitution of incoming and outgoing calls. If there is no match in the first number
substitution rule list, the voice gateway will match against other number substitution rule lists.
•
Number substitution on voice entities—The voice gateway substitutes the calling and called
numbers based on the number substitution rule lists bound to voice entities.
•
Number substitution on a specific subscriber line—The voice gateway substitutes the
calling and called numbers of incoming calls based on the number substitution rules configured
on the receiving line.
Configuration prerequisites
The required basic configurations have been completed on POTS, VoIP, and VoFR entities.
Configuration procedure
To configure global number substitution:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a number substitution
rule list and enter voice
number-substitute list-number
N/A
114
Step
Command
Remarks
number-substitute view.
Optional.
Configure a dot-match rule.
dot-match { end-only | left-right
| right-left }
6.
Configure a number
substitution rule.
rule rule-tag input-number
output-number [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *
7.
Configure the preferred
number substitution rule.
first-rule rule-number
By default, the preferred number
substitution rule is not configured.
8.
Exit voice number-substitute
view and enter voice dial
program view.
quit
N/A
Bind the calling/called
number of incoming/outgoing
calls to a number substitution
rule list.
substitute { incoming-call |
outgoing-call } { called | calling }
list-number
5.
9.
By default, the dot match rule is
end-only.
Optional.
By default, no number substitution
rule is configured.
Optional.
Optional.
By default, no number substitution
rule list is bound. That is to say, no
number substitution is performed.
To configure number substitution for a voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a number substitution
rule list and enter voice
number-substitute view.
number-substitute list-number
N/A
Configure a dot-match rule.
dot-match { end-only | left-right
| right-left }
6.
Configure a number
substitution rule.
rule rule-tag input-number
output-number [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *
7.
Configure the preferred
number substitution rule.
first-rule rule-number
By default, the preferred number
substitution rule is not configured.
8.
Exit voice number-substitute
view and enter voice dial
program view.
quit
N/A
Enter voice entity view.
entity entity-number { pots | vofr |
voip }
N/A
5.
9.
Optional.
By default, the dot match rule is
end-only.
Optional.
By default, no number substitution
rule is configured.
Optional.
115
Step
Command
10. Bind a number substitution
rule list to a voice entity.
substitute { called | calling }
list-number
Remarks
Optional.
By default, no number substitution
rule list is bound to a voice entity.
To configure number substitution for a voice subscriber line:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create a number substitution
rule list and enter voice
number-substitute view.
number-substitute list-number
N/A
Configure a dot-match rule.
dot-match { end-only | left-right
| right-left }
6.
Configure a number
substitution rule.
rule rule-tag input-number
output-number [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *
7.
Configure the preferred
number substitution rule.
first-rule rule-number
By default, the preferred number
substitution rule is not configured.
8.
Exit voice number-substitute
view and enter voice dial
program view.
quit
N/A
Exit voice dial program view
and enter voice view.
quit
N/A
10. Exit voice view and enter
system view.
quit
N/A
11. Enter voice subscriber line
view.
subscriber-line line-number
N/A
12. Configure a number
substitution rule list for a
subscriber line.
substitute { called | calling }
list-number
5.
9.
Optional.
By default, the dot match rule is
end-only.
Optional.
By default, no number substitution
rule is configured.
Optional.
Optional.
By default, no number substitution
rule list is configured for a
subscriber line.
IMPORTANT:
Whatever number substitution mode is configured, the voice gateway performs number substitution
only once on a given number.
Configuring a number sending mode
Configure a number sending mode to control how the originating gateway sends a called number.
Three number sending modes are available:
116
•
Send the least significant digits (configured by the send-number digit-number command) of a
called number.
•
Sends all digits of a called number.
•
Send a truncated called number. When the match-template command configured for a voice
entity contains an ending wildcard, only the digits that match the wildcard are sent.
Configuration prerequisites
The required basic configurations have been completed on POTS entities.
Configuration procedure
To configure a number sending mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter POTS voice entity
view.
entity entity-number pots
N/A
5.
Configure a number sending
mode.
send-number { digit-number | all |
truncate }
By default, the truncate mode is
used.
6.
Bind a number template to
the local voice entity.
match-template match-string
By default, no number template is
bound to the local or trunk voice
subscriber line.
Configuring a dial prefix
The following section provides information about configuring a dial prefix.
Configuration prerequisites
The required basic functions have completed on POTS and VoIP entities.
Configuration procedure
You can configure a prefix for dialed PSTN numbers. When a POTS entity originates a call, the dial
prefix will be added to the called number.
To configure a dial prefix:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter POTS entity view.
entity entity-number pots
N/A
117
Step
Command
Remarks
5.
Configure a dial prefix.
dial-prefix string
No dial prefix by default.
6.
Bind a number template to
the local voice entity.
match-template match-string
By default, no number template is
bound to the local or trunk voice
subscriber line.
Displaying and maintaining dial plan configuration
Task
Command
Display information of the
configured number substitution
rule lists.
display voice
number-substitute [ list-tag ] [ |
{ begin | exclude | include }
regular-expression ]
Display information of the
configured subscriber groups.
display voice subscriber-group
{ subscriber-group-list-tag | all } [ |
{ begin | exclude | include }
regular-expression ]
Remarks
Available in any view.
Dial plan configuration examples
This section provides dial plan configuration examples.
Configuring number substitution
Network requirements
As shown in Figure 32, a PBX to forms a local telephony network at place A and place B respectively.
The following requirements must be met:
•
These two local telephony networks communicate through two voice gateways. Subscribers in
one PBX network can make ordinary calls to remote subscribers in the other PBX network over
a VoIP network.
•
Configure two FXO trunk lines between each router and its PBX and enable hunt group to
realize trunk line backup.
•
There are a financial department, market department, and sales department at both place A
(area code 021) and place B (area code 010). A department at place A only needs to know the
telephone numbers of the local departments and the area code of place B when calling a
department at place B. For example, the financial department at place B can dial 3366 to call
the local market department. The financial department at place B can dial 0103366 to call the
market department at place A, and the caller identification displayed on the terminal at place A
is 0211234, that is, the area code of place B + telephone number of the financial department at
place B.
118
Figure 32 Network diagram
Configuration consideration
The PBX (calling side) at place B changes the called number to an intermediate number.
The PBX (called side) at place A changes the received intermediate number to a local number before
initiating the call.
Configuration procedure
The following configuration supports dial plan–based calls from place B to place A only.
1.
Configure Router B:
# Set the IP address of the Ethernet interface to 2.2.2.2.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 2.2.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Configure a number substitution rule list for called numbers of outgoing calls.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] number-substitute 21101
[RouterB-voice-dial-substitute21101] rule 1 0101688 0001
[RouterB-voice-dial-substitute21101] rule 2 0103366 0002
[RouterB-voice-dial-substitute21101] rule 3 0102323 0003
# Configure a number substitution rule list for calling numbers of outgoing calls.
[RouterB-voice-dial-substitute21101] quit
[RouterB-voice-dial] number-substitute 21102
[RouterB-voice-dial-substitute21102] rule 1 1688 0210001
[RouterB-voice-dial-substitute21102] rule 2 3366 0210002
[RouterB-voice-dial-substitute21102] rule 3 2323 0210003
[RouterB-voice-dial-substitute21102] quit
# Configure a VoIP voice entity to place A.
[RouterB-voice-dial] entity 10 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] substitute called 21101
[RouterB-voice-dial-entity10] substitute calling 21102
# Configure FXO trunk line 1/0.
[RouterB-voice-dial] entity 1010 pots
[RouterB-voice-dial-entity1010] match-template ....
119
[RouterB-voice-dial-entity1010] line 1/0
[RouterB-voice-dial-entity1010] send-number all
# Configure FXO trunk line 1/1.
[RouterB-voice-dial-entity1010] quit
[RouterB-voice-dial] entity 2010 pots
[RouterB-voice-dial-entity2010] match-template ....
[RouterB-voice-dial-entity2010] line 1/1
[RouterB-voice-dial-entity2010] send-number all
# Enable hunt group.
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] hunt-group enable
[RouterB-subscriber-line1/0] quit
[RouterB] subscriber-line 1/1
[RouterB-subscriber-line1/1] hunt-group enable
2.
Configure Router A:
# Set the address of an Ethernet interface to 1.1.1.1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 1.1.1.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure a number substitution rule list for called numbers of incoming calls.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] number-substitute 101
[RouterA-voice-dial-substitute101] rule 1 ^0001$ 1234
[RouterA-voice-dial-substitute101] rule 2 ^0002$ 6788
[RouterA-voice-dial-substitute101] rule 3 ^0003$ 6565
# Configure a number substitution rule list for calling numbers of incoming calls.
[RouterA-voice-dial-substitute101] quit
[RouterA-voice-dial] number-substitute 102
[RouterA-voice-dial-substitute102] dot-match left-right
[RouterA-voice-dial-substitute102] rule 1 ^...0001$ ...1234
[RouterA-voice-dial-substitute102] rule 2 ^...0002$ ...6788
[RouterA-voice-dial-substitute102] rule 3 ^...0003$ ...6565
[RouterA-voice-dial-substitute102] quit
# Configure number substitution rules.
[RouterA-voice-dial] substitute incoming-call called 101
[RouterA-voice-dial] substitute incoming-call calling 102
# Configure FXO trunk line 1/0.
[RouterA-voice-dial] entity 1010 pots
[RouterA-voice-dial-entity1010] match-template ....
[RouterA-voice-dial-entity1010] line 1/0
[RouterA-voice-dial-entity1010] send-number all
# Configure FXO trunk line 1/1.
[RouterA-voice-dial-entity1010] quit
[RouterA-voice-dial] entity 2010 pots
[RouterA-voice-dial-entity2010] match-template ....
120
[RouterA-voice-dial-entity2010] line 1/1
[RouterA-voice-dial-entity2010] send-number all
# Enable hunt group.
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] hunt-group enable
[RouterB-subscriber-line1/0] quit
[RouterB] subscriber-line 1/1
[RouterB-subscriber-line1/1] hunt-group enable
Configuring the match order for voice entity selection
Network requirements
As shown in Figure 33, two telephones connected to Router A can make PSTN calls through two
trunk voice gateways (Router B and Router C) and the trunk lines of these two trunk voice gateways
should be fully utilized.
Figure 33 Network diagram
Eth2/1
1.1.1.3/24
FXO Line 1/0
FXO Line 1/1
010-1234
Router A
Router C
WAN
PBX
010-1235
1000
Eth2/1
1.1.1.1/24
FXO Line 1/0
FXO Line 1/1
Eth2/1
1001
Router B 1.1.1.2/24
Configuration consideration
The select-rule rule-order 1 4 command can implement load sharing. Because the first rule "exact
match" cannot distinguish the priority between Router B and Router C, Router A will use the fourth
rule "longest idle time" to make sure that the resources of the two gateways are fully, equally utilized.
Configuration procedure
1.
Configure Router A:
# Configure an Ethernet address.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 1.1.1.1 24
[RouterA-Ethernet2/1] quit
# Configure VoIP entities that respectively originate calls to Router B and Router C.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 010....
[RouterA-voice-dial-entity2000] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity2001 voip
[RouterA-voice-dial-entity2001] match-template 010....
[RouterA-voice-dial-entity2001] address sip ip 1.1.1.3
[RouterA-voice-dial-entity2001] quit
121
# Configure POTS entities.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial] entity1001 pots
[RouterA-voice-dial-entity1001] match-template 1001
[RouterA-voice-dial-entity1001] line 1/1
[RouterA-voice-dial-entity1001] quit
# Configure rules in the match order for voice entity selection.
[RouterA-voice-dial] select-rule rule-order 1 4
2.
Configure Router B:
# Configure an Ethernet address.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 1.1.1.2 24
[RouterB-Ethernet2/1] quit
# Configure POTS entities.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 pots
[RouterB-voice-dial-entity1000] match-template 010....
[RouterB-voice-dial-entity1000] line 1/0
[RouterB-voice-dial-entity1000] send-number all
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] entity1001 pots
[RouterB-voice-dial-entity1001] match-template 010....
[RouterB-voice-dial-entity1001] line 1/1
[RouterB-voice-dial-entity1001] send-number all
[RouterB-voice-dial-entity1001] quit
# Configure rules in the match order for voice entity selection.
[RouterB-voice-dial] select-rule rule-order 1 4
3.
Configure Router C:
# Configure an Ethernet address.
<RouterC> system-view
[RouterC] interface ethernet 2/1
[RouterC-Ethernet2/1] ip address 1.1.1.3 24
[RouterC-Ethernet2/1] quit
# Configure POTS entities.
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 1000 pots
[RouterC-voice-dial-entity1000] match-template 010....
[RouterC-voice-dial-entity1000] line 1/0
[RouterC-voice-dial-entity1000] send-number all
[RouterC-voice-dial-entity1000] quit
[RouterC-voice-dial] entity1001 pots
[RouterC-voice-dial-entity1001] match-template 010....
[RouterC-voice-dial-entity1001] line 1/1
122
[RouterC-voice-dial-entity1001] send-number all
[RouterC-voice-dial-entity1001] quit
# Configure rules in the match order for voice entity selection
[RouterC-voice-dial] select-rule rule-order 1 4
Configuring the maximum-call-connection set
Network requirement
As shown in Figure 34, there are one trunking voice gateway (Router C) and two subscriber voice
gateways (Router A and Router B) in a city. To prevent the trunk lines from being totally occupied by
either subscriber voice gateway, you must restrict the number of calls originated from Router A and
Router B respectively.
Figure 34 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure an Ethernet address.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 1.1.1.1 24
[RouterA-Ethernet2/1] quit
# Configure a VoIP entity.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 010....
[RouterA-voice-dial-entity2000] address sip ip 1.1.1.3
[RouterA-voice-dial-entity2000] quit
# Configure POTS entities.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] entity1001 pots
[RouterA-voice-dial-entity1001] match-template 1001
[RouterA-voice-dial-entity1001] line 1/1
[RouterA-voice-dial-entity1001] quit
123
# Configure the maximum-call-connection set.
[RouterA-voice-dial] max-call 1 2
# Bind the maximum-call-connection set to a voice entity.
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] max-call 1
2.
Configure Router B:
# Configure an Ethernet address.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 1.1.1.2 24
[RouterB-Ethernet2/1] quit
# Configure a VoIP entity.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] match-template 010....
[RouterB-voice-dial-entity1000] address sip ip 1.1.1.3
[RouterB-voice-dial-entity1000] quit
# Configure POTS entities.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] quit
[RouterB-voice-dial] entity2001 pots
[RouterB-voice-dial-entity2001] match-template 2001
[RouterB-voice-dial-entity2001] line 1/1
[RouterB-voice-dial-entity2001] quit
# Configure the maximum-call-connection set.
[RouterB-voice-dial] max-call 1 2
# Bind the maximum-call-connection set to a voice entity.
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] max-call 1
3.
Configure Router C:
# Configure an Ethernet address.
<RouterC> system-view
[RouterC] interface ethernet 2/1
[RouterC-Ethernet2/1] ip address 1.1.1.3 24
[RouterC-Ethernet2/1] quit
# Configure POTS entities.
[RouterC-voice-dial] entity 1000 pots
[RouterC-voice-dial-entity1000] match-template 010....
[RouterC-voice-dial-entity1000] line 5/0
[RouterC-voice-dial-entity1000] send-number all
[RouterC-voice-dial-entity1000] quit
[RouterC-voice-dial] entity1001 pots
[RouterC-voice-dial-entity1001] match-template 010
[RouterC-voice-dial-entity1001] line 5/1
124
[RouterC-voice-dial-entity1001] send-number all
[RouterC-voice-dial] entity1002 pots
[RouterC-voice-dial-entity1002] match-template 010....
[RouterC-voice-dial-entity1002] line 5/2
[RouterC-voice-dial-entity1002] send-number all
[RouterC-voice-dial-entity1001] quit
[RouterC-voice-dial-entity1002] quit
[RouterC-voice-dial] entity1003 pots
[RouterC-voice-dial-entity1003] match-template 010....
[RouterC-voice-dial-entity1003] line 5/3
[RouterC-voice-dial-entity1003] send-number all
[RouterC-voice-dial-entity1003] return
# Enable hunt group for voice subscriber lines.
<RouterC> system-view
[RouterC] subscriber-line 1/0
[RouterC-subscriber-line1/0] hunt-group enable
[RouterC-subscriber-line1/0] quit
[RouterC] subscriber-line 1/1
[RouterC-subscriber-line1/1] hunt-group enable
[RouterC-subscriber-line1/1] quit
[RouterC] subscriber-line 1/2
[RouterC-subscriber-line1/2] hunt-group enable
[RouterC-subscriber-line1/2] quit
[RouterC] subscriber-line 1/3
[RouterC-subscriber-line1/3] hunt-group enable
Configuring call authority control
Network requirements
As shown in Figure 35, Router A, Router B, and Router C are located at place A, place B, and place
C, respectively, and they are all connected to the SIP server to allow subscribers to make SIP calls.
When VoIP links fail for some reason, PSTN links that provide backup for VoIP links can be
automatically brought up. Subscribers whose telephone numbers begin with 1100 at place A can
originate calls to place B while subscribers whose telephone number begin with 1200 can originate
calls to both place B and place C.
125
Figure 35 Network diagram
Configuration procedure
This example does not provide SIP server and digital subscriber line configurations. For more
information, see "Configuring SIP" and "Configuring voice subscriber lines."
1.
Configure Router A:
# Configure two subscriber groups.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] subscriber-group 1
[RouterA-voice-dial-group1] match-template 1100..
[RouterA-voice-dial-group1] quit
[RouterA-voice-dial] subscriber-group 2
[RouterA-voice-dial-group2] match-template 1200..
[RouterA-voice-dial-group2] quit
# Configure VoIP entities for place B and place C.
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip proxy
[RouterA-voice-dial-entity2000] match-template 2...
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip proxy
[RouterA-voice-dial-entity3000] match-template 3...
[RouterA-voice-dial-entity3000] quit
# Configure a POTS entity for place B.
[RouterA-voice-dial] entity 2100 pots
[RouterA-voice-dial-entity2100] line 1/0:15
[RouterA-voice-dial-entity2100] send-number all
[RouterA-voice-dial-entity2100] match-template 2...
[RouterA-voice-dial-entity2100] caller-group permit 1
[RouterA-voice-dial-entity2100] caller-group permit 2
[RouterA-voice-dial-entity2100] quit
126
# Configure a POTS entity for place C.
[RouterA-voice-dial] entity 3100 pots
[RouterA-voice-dial-entity3100] line 2/0:15
[RouterA-voice-dial-entity3100] send-number all
[RouterA-voice-dial-entity3100] match-template 3...
[RouterA-voice-dial-entity3100] caller-group permit 2
2.
Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2100 pots
[RouterB-voice-dial-entity2100] line 1/0:15
[RouterB-voice-dial-entity2100] send-number all
[RouterB-voice-dial-entity2100] match-template 2...
3.
Configure Router C:
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3100 pots
[RouterC-voice-dial-entity3100] line 1/0:15
[RouterC-voice-dial-entity3100] send-number all
[RouterC-voice-dial-entity3100] match-template 3...
4.
Display information of the configured subscriber groups:
<RouterA> display voice subscriber-group all
Current configuration of subscriber group 1
#
Description : <NULL>
Referenced by entities:
Type: POTS
Tag: 2100
Include match templates:
Match-template: 1100..
#
END
Current configuration of subscriber group 2
#
Description : <NULL>
Referenced by entities:
Type: POTS
Tag: 2100
Type: POTS
Tag: 3100
Include match templates:
Match-template: 1200..
#
END
127
Configuring SIP
Overview
The Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify,
and terminate multimedia sessions such as IP phone calls, multimedia session, and multimedia
conferences. It is the core component in the multimedia data and control architecture of the IETF
(RFC 3261).
SIP is responsible for signaling control in IP networks and communication with soft switch platforms.
The intent is to build a next-generation value-added service platform to deliver better value-added
services to telecom carriers, banks, and financial organizations.
SIP is used for initiating sessions. It sets up and terminates a multimedia session involving a group of
participants and dynamically adjusts and modifies session characteristics such as required session
bandwidth, media type (voice, video, or data), media encoding/decoding format, and
multicast/unicast. SIP is based on text encoding and constructed by using the mature protocol HTTP
as a model. Easy to extend and implement, it is suitable for implementing Internet-based multimedia
conference systems.
SIP adopts the client/server model and sets up user calls through communication between user
agents and proxy servers.
Terminology
Multimedia session
According to RFC2327, "A multimedia session is a set of multimedia senders and receivers and the
data streams flowing from senders to receivers. A multimedia conference is an example of a
multimedia session."
A session is identified by a set of username, session ID, network type, address type, and address.
User agent
A user agent (UA), or a SIP endpoint, is a SIP-enabled multimedia session endpoint, which can be a
phone, a gateway, or a router. Usually, a SIP-enabled router serves as a SIP UA.
There are two types of UAs: user agent client (UAC) and user agent server (UAS). To make a voice
call, a SIP endpoint needs to process a SIP request as a UAS and initiate a SIP request as a UAC.
A UAC is a device that initiates a session request. It can be a calling SIP endpoint or a proxy server
forwarding a request to a called endpoint for example.
A UAS is a device that generates a response to a SIP request. It can be a called SIP endpoint or a
proxy server receiving a request from a calling endpoint for example.
Proxy server
A proxy server is a device that forwards session requests to a called UA on behalf of a calling UA (a
SIP endpoint) and responds to the calling UA on behalf of the called UA.
When the proxy server receives a request from a calling UA, it first requests its location server for
information on called UA location and call policies of calling UA and called UA. If the location
information of the called UA is available and the calling UA is allowed to make the call, the proxy
server then forwards the request to the called UA.
Redirect server
A redirect server sends a new connection address to a requesting client.
128
For example, when receiving a request from a calling UA, the redirect server searches for the
location information of the called UA and returns the location information to the UA. This location can
be that of the called UA or another proxy server, to which the UA can initiate the session request
again. The subsequent procedure is the same as that for calling a called UA directly or for calling a
proxy server.
Location server
A location server is a device that provides UA information to proxy and redirect servers; it retains UA
information received by a registrar. The location server and registrar can located on the same device
as two logical components or located on different devices.
Registrar
A registrar receives UAs’ registrations. The registration information (for example, the local telephone
number) is usually stored on the location server for future retrieval. The location server and the
registrar are both logical components and are usually co-located.
SIP functions and features
Functions
SIP supports the following facets of establishing and terminating multimedia communications:
•
Locating called SIP endpoints, the most powerful function of SIP. For this purpose, SIP can use
the registration information of SIP endpoints on the registrar. In addition, it can enhance its user
location service by using other location services provided by the domain name server (DNS)
and lightweight directory access protocol (LDAP).
•
Determining user availability, making sure whether a called endpoint can participate in a
session. SIP supports multiple address description and addressing styles, including SIP-URI
(for example, SIP: 123456@172.18.24.11), Tel-URL (for example, Tel: +1312000), and
SIPS-URI (SIPS: 123456@172.18.24.11). Thus, a SIP caller can identify whether a callee is
attached to a PSTN network by callee's address, and then initiate and set up the call to the
callee through the gateway connected to the PSTN.
•
Determining user capabilities, that is, the media type and media parameters of a called
endpoint. In a message exchange process, each SIP endpoint sends such information in
transmitted messages so that all other participants can learn about its capabilities.
•
Setting up a session, or session parameters, at both callee and caller sides. Two parties can
select the appropriate capabilities for session setup through negotiation about media type and
media parameters to be used.
•
Managing sessions by modifying session parameters or terminating sessions.
Features
SIP delivers the following features:
•
Open standards—Can accommodate new functions, products, and services introduced by
different service providers.
•
Flexible configuration—Accommodates a wide range of dialup, wire, and wireless devices,
allows highly flexible configurations, and can work with other systems.
•
Scalable system—Allows expansion as enterprises grow.
•
Support to remote users—With SIP, an enterprise network can extend to all its users,
wherever they are.
•
Competitive advantage potentials—More SIP-based services are emerging.
•
Consistent communication method—Management becomes easier as the result of
consistency in dialup mode and system access method used by branches, SOHOs, and
traveling personnel.
129
•
Quick launch—The system can be updated quickly to accommodate new branches and
personnel, as well as changes resulting from job rotation or relocation.
•
Easy to install and maintain—Even non-professional individuals can install and maintain SIP
systems.
SIP messages
SIP messages, consisting of SIP request messages and SIP response messages, are encoded in
text mode.
SIP request messages include INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER. RFC 3261
defines the following six request messages:
•
INVITE—Used to invite a user to join a call.
•
ACK—Used to acknowledge the response to a request.
•
OPTIONS—Used to query for the capabilities.
•
BYE—Used to release an established call.
•
CANCEL—Used to give up a call attempt.
•
REGISTER—Used to register with the SIP registrar.
SIP response messages, used to respond to SIP requests, indicate the status of a call or registration,
whether succeeded or failed. Response messages are distinguished by status codes. As shown
in Table 13, each status code is a 3-digit integer, where the first digit defines the class of a response,
and the last two digits describe the response message in more detail.
Table 13 Status codes of response messages
Code
Description
Class
100–199
The request is received and is being processed.
Provisional
200–299
The request is successfully received, understood, and accepted.
Success
300–399
A further action needs to be taken in order to process the request.
Redirection
400–499
The request contains bad syntax and thus cannot be processed.
Client error
500–599
The request cannot be processed due to UAS or server error.
Server error
600–699
The request cannot be processed by any UAS or server.
Global error
SIP fundamentals
Registration
In a complete SIP system, all SIP endpoints working as UAs should register with SIP registrars,
providing information such as location, session capabilities, and call policy.
As shown in Figure 36, a SIP UA sends its registrar a REGISTER request at startup or in response to
an administrative registration operation, carrying all the information that must be recorded. Upon
receipt of the request, the registrar sends back a response notifying receipt of the request, and a 200
OK (SUCCESS) message if the registration is accepted.
130
Figure 36 Message exchange for a UA to register with a Registrar
Call setup
SIP operates in the Client/Server mode and sets up calls through communication between UA and
proxy server.
Figure 37 Network diagram
In the above figure, Telephone A will call Telephone B, and Router A and Router B work as SIP
endpoints (UAs).
The following is the procedure for connecting a call from Telephone A to Telephone B:
1.
Telephone A sends the number of Telephone B.
2.
Upon receipt of the call, Router A sends a session request (INVITE) to the proxy server.
3.
The proxy server consults its database for information corresponding to the number of
Telephone B. If such information is available, it forwards the request to Router B.
4.
Router B, after receiving the request, responds to the proxy server and makes Telephone B ring
if Telephone B is available.
5.
The proxy server forwards the response to Router A. The response discussed here includes
two provisional response messages (100 Trying and 180 Ringing) and one success response
(200 OK).
Figure 38 illustrates the complete call setup procedure.
131
Figure 38 Call setup procedures involving a proxy server
This is a simplified scenario where only one proxy server is involved and no registrar is present. A
complex scenario, however, might involve multiple proxy servers and registrars.
Call redirection
As shown in Figure 39, when a SIP redirect server receives a session request, it sends back a
response indicating the address of the called SIP endpoint instead of forwarding the request. The
calling and called endpoints send request and response to each other directly.
132
Figure 39 Call redirection procedure for UAs
Internet
User agent
User agent
Redirect Server
INVITE
100 Trying
302 Moved Temporarily
ACK
INVITE
100 Trying
200 OK
ACK
This is a common application. Fundamentally, a redirect server can respond with the address of a
proxy server as well. The subsequent call procedures are the same as the call procedures involving
proxy servers.
Support for transport layer protocols
As an application layer protocol, SIP supports the following transport layer protocols:
•
UDP—UDP is a connectionless protocol and does not provide reliability; therefore, SIP
connections established over UDP are unreliable.
•
TCP—Ensures transmission reliability for SIP messages. TCP provides connection-oriented
and reliable transmission for SIP-based VoIP communications. Using TCP, SIP need not
consider packet loss and retransmission issues.
•
Transport layer security (TLS)—Ensures transmission security for SIP messages. For more
information, see "Signaling encryption."
These transport layer protocols have their own benefits, enabling you to select a protocol based on
your network environment. The system does not support transport layer protocol switchover during
communication.
SIP security
Signaling encryption
TLS runs over TCP and provides a complete set of authentication and encryption solutions for
application layer protocols. When establishing a TLS connection, both sides need to authenticate
each other by using their own digital certificates and can communicate with each other only after
133
passing authentication. SIP messages are encrypted during SIP over TLS transmissions to prevent
your data from being sniffed. This increases the security of voice communications.
For more information about signaling encryption, see "Configuring TLS for SIP sessions."
SIP over TLS requires the configuration of TLS security policies. For information about how to
configure the TLS security policies, see Security Configuration Guide.
Media flow encryption
Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) are supported
media flow protocols. RTP provides end-to-end real-time transmission for real-time data such as
audio and video data. RTCP monitors data transmission in real time and performs congestion and
traffic control in time. RTP and RTCP can work together to optimize the transmission efficiency by
providing efficient replies and minimizing overheads.
Media flows are transmitted in plain text. To ensure transmission security, the Secure Real-Time
Transport Protocol (SRTP) was introduced.
SRTP provides for encryption of the RTP/RTCP packet payload, for authentication of the entire
RTP/RTCP packet, and for packet replay protection. For more information about media flow
encryption, see "Configuring media flow protocols for SIP calls."
The first step of SRTP encryption is to negotiate encryption information, which can only be carried in
the crypto header field of the Session Description Protocol (SDP) at present. The initiator sends its
encryption information to the receiver for negotiation. If the negotiation is successful, the receiver
returns corresponding encryption information. After a session is established, each end uses its own
key to encrypt sent RTP/RTCP packets and uses the key of the peer to decrypt received RTP/RTCP
packets.
As shown in Table 14, SDP negotiation includes the following cryptographic attributes:
Table 14 Cryptographic attributes
Attribute
Description
Remarks
Tag
The tag attribute is an identifier for a particular cryptographic
attribute to determine which of the several offered
cryptographic attributes was chosen by the receiver.
Required.
Crypto-Suite
The crypto-suite attribute defines the encryption and
authentication algorithm. The device supports suites
AES_CM_128_HMAC_SHA1_80 and
AES_CM_128_HMAC_SHA1_32.
Required.
Key Parameters
The key parameters attribute defines key information,
including the key generation algorithm and the key value.
Required.
Session
Parameters
The session parameters attribute defines session parameters,
such as key generation rate, UNENCRYPTED_SRTP,
UNENCRYPTED_SRTCP, UNAUTHENTICATED_SRTP, and
FEC.
Optional.
Not supported.
When SRTP is used to encrypt RTP/RTCP packets, the encryption engine, if enabled, encrypts and
authenticates RTP/RTCP packets. If the encryption engine is disabled, the CPU encrypts and
authenticates RTP/RTCP packets. For more information about the encryption engine, see Security
Configuration Guide.
SRTP is available only for SIP calls. SIP trunk devices do not support SRTP.
134
TLS-SRTP combinations
TLS protects control signaling, and SRTP encrypts and authenticates voice media flows. You can
use them separately or together. As shown in Table 15, there are four combinations of TLS and
SRTP.
Table 15 TLS-SRTP combinations
TLS
SRTP
On
On
Description
Signaling packets are secured. Personal information is protected.
Media packets are secured. Call conversations are protected.
Recommended.
Off
On
On
Off
Off
Off
Signaling packets are not secured. Personal information is not protected.
Media packets are secured. Call conversations are protected.
Signaling packets are secured. Personal information is protected.
Media packets are not secured. Call conversations are not protected.
Signaling packets are not secured. Personal information is not protected.
Media packets are not secured. Call conversations are not protected.
Support for basic QSIG call
As shown in Figure 40, the two ISDN networks connected through a SIP network need to
communicate with each other by using QSIG signaling. To support this application, the voice
gateways Router A and B need to translate between QSIG and SIP signaling messages. However,
this type of gateways can only translate basic call signaling rather than a great amount of QSIG
signaling related to supplementary services. To solve this problem, a method of encapsulating
original QSIG signaling messages within SIP messages was introduced to tunnel ISDN signaling
over a SIP network. This method is called SIP for telephones (SIP-T).
Figure 40 Tunneling QSIG signaling messages over a SIP network
The core of the STP-T protocol is to encapsulate and decapsulate ISDN messages. The ingress
gateway encapsulates QSIG signaling messages received from a sender into SIP messages, and
sends them to the egress gateway. Upon receiving SIP messages, the egress gateway extracts the
QSIG signaling messages from the SIP messages and sends them to the receiving end.
Compared with the simple mapping between QSIG signaling and SIP signaling, the SIP-T protocol
enhances the mapping between the two signaling protocols by improving the integrity and
abundance of the ISDN signaling contents. This enables the lossless transmission of ISDN signaling
over the SIP network between two ISDN networks.
The device only supports encapsulating QSIG messages within SIP messages.
135
For configuration of basic QSIG call supported by the SIP-T protocol, see "Configuring digital voice
subscriber lines."
VRF-aware SIP
VRF-aware SIP enables telephones to call each other in an L3VPN.
As shown in Figure 41, two VPNs Voice and Data exist on the network. The two VPNs cannot
access each other. After you specify the Voice VPN instance on PE 1, Telephone 3 can make SIP
calls with telephones 1 and 2 in the VPN Voice.
Figure 41 VRF-aware SIP
Hardware compatibility with SIP
SIP is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
SIP configuration task list
Complete the following tasks to configure SIP:
Task
Configuring SIP
Remarks
Configuring SIP authentication information
136
Optional.
Task
UA registration
Remarks
Configuring registrar information for a SIP UA
Required.
Configuring proxy server information for a SIP UA
Optional.
Configuring registration timers
Optional.
Configuring call failure-triggered re-registration
Optional.
Configuring fuzzy telephone number registration
Optional.
Enabling SIP registration function
Required.
Configuring SIP server keepalive and backup
Optional.
Configuring SIP routing
Required.
Configuring user information
Optional.
Configuring outbound SIP proxy server information for a SIP UA
Optional.
Configuring transport layer protocol for SIP calls
Required when TCP
or TLS is specified
as the transport layer
protocol.
Configuring SIP
extensions
Strict SIP routing
The feature is
enabled by default,
and requires no
configuration.
Configuring out-of-band SIP DTMF transmission mode
Optional.
Configuring source IP address binding for SIP messages
Optional.
Configuring a domain name for the SIP UA
Optional.
Configuring SIP compatibility
Optional.
Configuring user-agent and server header fields
Optional.
Configuring SIP extensions for caller identity and privacy
Optional.
Configuring call release cause code mapping
Optional.
Configuring periodic refresh of SIP sessions
Optional.
Enabling early media negotiation
Optional.
Configuring VRF-aware SIP
Optional.
Configuring SIP UA registration
UAs function to register and to initiate session requests or make responses.
A UA registers with the registrar by sending a REGISTER request containing such information as
address, route, and number. Therefore, when it is called by some other UA through a proxy server,
the proxy server can consult the registrar for its registration information.
The basic functions that a UA must have are the ability to initiate session requests as a caller and the
ability to make responses as a callee. After a user attached to a UA-supported router working as UA
dials a number, the router sends a session request to the called UA as a caller either directly or
through one or more proxy servers.
137
Configuration prerequisites
The configuration at the server end is completed.
Configuring SIP authentication information
SIP authentication information is required in the following two cases:
•
UAs use a proxy server and the proxy server needs to authenticate UAs.
•
The peer SIP device needs to authenticate the local device during SIP message exchange.
Authentication information selection rule
If the authentication information is configured with the user command, POTS entity view is preferred.
Otherwise, the authentication information configured in SIP client view is used.
If realm is configured on a SIP UA, make sure the value is the same as that configured on the server.
Otherwise, the SIP UA will fail the authentication due to mismatch. If realm is not configured on the
SIP UA, the SIP UA will perform no realm match and consider that the value of realm configured on
the server is trusted.
If it is necessary to configure authentication information in POTS entity view or IVR entity view, the
same authentication information is recommended for the POTS entities or IVR entities configured
with the same telephone number.
IMPORTANT:
In the case of authentication, it is forbidden to execute the user command after the registration
function is enabled because this operation might result in a registration update failure.
The configuration on the SIP server is completed.
Configuration procedure
To configure global SIP authentication information:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure global SIP
authentication
information.
user username password
{ cipher | simple } password
[ cnonce cnonce | realm realm ] *
Optional.
By default, the SIP username is
VOICE-GATEWAY and the
authentication password is VOICE-SIP
in SIP client view.
To configuring SIP authentication information in POTS or IVR entity view:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { ivr | pots }
N/A
5.
Configure SIP authentication
user username password
Optional.
138
Step
information in POTS entity
view or IVR entity view.
Command
Remarks
{ cipher | simple } password
[ cnonce cnonce | realm realm ] *
By default, no SIP authentication
information is configured in POTS
entity view or IVR entity view.
Configuring registrar information for a SIP UA
Configuration guidelines
The transport layer protocol specified in the registrar command must have been specified with the
listen transport command. Otherwise, no register request can be initiated.
If TLS is specified in the registrar command, you also need to configure the secure sockets layer
(SSL) policy name of the client with the crypto command. Otherwise, no register request can be
initiated.
You can use the registrar command only when the SIP registration function is disabled.
Configuration procedure
To configure registrar information for a SIP UA:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure the registrar
information for the SIP
UA.
registrar { dns domain-name | ipv4
ip-address } [ port port-number ]
[ expires seconds ] [ tcp | tls ]
[ scheme { sip | sips } ] [ slave ]
By default, no registrar
information is configured for a SIP
UA.
If the domain name of the registrar
is specified, DNS lookup will be
performed as configured. For
more information, see
"Configuration procedure (3:
destination domain)."
Configuring proxy server information for a SIP UA
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
By default, no proxy server
information is configured for a SIP
UA.
4.
Configure the proxy server
information for the SIP UA.
proxy { dns domain-name | ipv4
ip-address } [ port port-number ]
139
If the domain name of the proxy
server is specified, DNS lookup
will be performed as configured.
For more information, see
"Configuration procedure (3:
destination domain)."
Configuring registration timers
A voice entity or an SIP trunk account expires after it has registered with the registrar for a specified
period of time, which is the registration expiration interval.
To ensure the validity of registration information of a voice entity or an SIP trunk account on the
registrar, the voice entity or SIP trunk account must re-register with the registrar at a specified time
before the registration expiration interval is reached. Use the timer registration divider or timer
registration threshold command to set the time when the voice entity or SIP trunk account
re-registers with the registrar.
•
Use the timer registration divider command to set the expiration percentage. When the time,
which is registration expiration interval multiplied by expiration percentage, is reached, the
voice entity or SIP trunk account re-registers with the registrar.
•
Use the timer registration threshold command to set the lead time before expiration. When
the time, which is registration expiration interval minus lead time before expiration, is reached,
the voice entity or SIP trunk account re-registers with the registrar.
You can configure both timers. In this case, the actual re-registration time is decided by the timer that
expires first. In other words, the voice entity or SIP trunk account tries to re-register with the registrar
when any one of the two timers expires.
Configuration guidelines
Registration timers are available to SIP trunk accounts. For more information about the SIP trunk
account, see "Configuring a SIP trunk account."
You can specify the registration expiration interval by providing the expires keyword in the registrar
or registrar server-group command. Otherwise, the registration expiration interval specified in the
timer registration expires command takes effect. The proxy command does not include the
expires keyword, so when registrations are initiated from a voice entity to the registrar specified by
the proxy command, the registration expiration interval is determined by only the timer registration
expires command.
Configuration procedure
To configure registration timers:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Set the interval for the voice
entity or SIP trunk account to
re-register with the registrar
after registration failure.
timer registration retry seconds
Set the registration
expiration time.
timer registration expires
seconds
Optional.
Set the registration
percentage.
timer registration divider
percentage
Optional.
Set the lead time before
registration.
timer registration threshold
seconds
Optional.
5.
6.
7.
140
Optional.
The default is 240 seconds.
The default is 3600 seconds.
The default is 80.
The default is 0.
Configuring call failure-triggered re-registration
If the registrar fails and the registration information is lost after the registrar reboots, a service (such
as a call) initiated by a previously registered number will fail and the registrar will return an error code.
If the failure-triggered re-registration is enabled and the error code is 5xx (except for 502, 504, 505,
and 513), 403, or 408, the device will re-register the number on the registrar.
To configure call failure-triggered re-registration:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Enable call failure-triggered
re-registration.
call-fallback register
Optional.
Not enabled by default.
Configuring fuzzy telephone number registration
Fuzzy telephone number registration refers to the use of a wildcard (including the dot . and the
character T), rather than a standard E.164 number in the match template of a POTS entity.
After enabling fuzzy telephone number registration, the voice gateway (router) retains dots and
substitutes asterisks (*) for Ts when sending REGISTER messages.
To configure fuzzy telephone number registration:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Enable fuzzy (wildcard)
telephone number
registration.
wildcard-register enable
Disabled by default.
NOTE:
To use the fuzzy telephone number registration function, make sure the registrar and the location
server also support the function.
Enabling SIP registration function
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Enable the SIP registration
function.
register-enable on
Disabled by default.
5.
Enable checking the status
of voice subscriber lines
line-check enable
Optional.
141
Step
Command
associated with POTS voice
entities.
Remarks
By default, before registering
numbers for a POTS voice entity,
the device checks the status of the
voice subscriber line associated
to the POTS voice entity. The
device can send REGISTER
requests for numbers only when
the status of the line is up.
Configuring SIP server keepalive and backup
The keepalive function detects whether a SIP server is available in either of the following two ways:
•
Options keepalive—The device periodically sends Options packets to detect the availability of
the SIP server. If the response from the server is 408 or 5xx (except 502, 504, 505, and 513),
the server is unavailable.
•
Register keepalive—The register packets can also detect the availability of the SIP server. If
the response from the server is 408 or 5xx (except 502, 504, 505, and 513), the server is
unavailable.
Two backup modes are available in SIP client view.
•
Parking backup mode—If the current server is unavailable, the device sends options or
register packets to a backup server, and will not switch back after the original server recovers.
Before enabling the parking backup mode, enable the options or register keepalive with the
keepalive command.
•
Homing backup mode—The device sends Options packets to both the current server and the
server with the highest priority, regardless of the state of the server with the highest priority. If
the current server fails, the backup server becomes the current server. If the server with the
highest priority is available, the current server changes from the backup server to the server
with the highest priority. Before enabling homing backup, enable options keepalive with the
keepalive command.
To configure SIP server keepalive and backup:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure the keepalive
function.
keepalive { options [ interval
seconds ] | register }
Register keepalive is used by
default.
5.
Specify the backup mode.
redundancy mode { homing |
parking }
Parking backup mode is used by
default.
Configuring SIP routing
The following SIP routing policies are available on your router:
•
Use the destination IP address. Configure the IP address and port number of the destination.
•
Use the proxy server. If a SIP server is present, use the SIP proxy server for SIP message
interaction.
142
•
Use the domain name as the destination address. Without the need of obtaining the IP address
of the destination, the client only needs to know the unique domain name of the destination on
the network, and communicates with the destination through DNS.
•
Use the E.164 Number to URI Mapping (ENUM). ENUM translates telephone numbers into
Internet addresses.
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuration procedure (1: destination IP address)
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP entity view.
entity entity-number voip
N/A
5.
Configure the destination IP
address.
address sip ip ip-address [ port
port-number ]
Not configured by default.
For more information about DNS, see Layer 3—IP Services Configuration Guide.
Configuration procedure (2: SIP proxy server)
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP entity view.
entity entity-number voip
N/A
5.
Enable the SIP proxy server.
address sip proxy
Not configured by default.
Configuration procedure (3: destination domain)
If the destination address is a domain name, the device performs DNS lookup by using either of the
following modes:
•
Type-A—If you use the dns-type command to set the DNS lookup mode as a-record, or both
the domain name and destination port number are configured, the device uses the Type-A
mode to perform DNS lookup.
•
SRV—If the domain name is configured but the destination port number is not, use the
dns-type command to set the DNS lookup mode to SRV.
To configure SIP routing (3: destination domain):
Step
Command
Remarks
1.
system-view
N/A
Enter system view.
143
Step
Command
Remarks
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
Sip
N/A
Optional.
The default DNS lookup method is
a-record.
If you configure the destination
port by using the address sip
{ dns domain-name [ port
port-number ] | enum-group
group-number }, proxy dns
domain-name [ port
port-number ], or mwi-server dns
domain-name [ port port-number ]
command, the DNS lookup mode
can only be Type-A.
4.
Set the DNS lookup mode.
dns-type { a-record | srv }
5.
Exit the SIP view.
quit
N/A
6.
Enter voice dial program
view.
dial-program
N/A
7.
Enter VoIP entity view.
entity entity-number voip
N/A
8.
Configure the domain name.
address sip dns domain-name
[ port port-number ]
Not configured by default.
For more information about DNS, see Layer 3—IP Services Configuration Guide.
Configuration procedure (4: ENUM)
With the ENUM feature, the device translates the destination number into the corresponding URL
according to the matching ENUM translation rule, and uses this URL to perform DNS lookup.
To configure SIP routing (4: ENUM):
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter ENUM rule group view.
enum-group group-number
N/A
5.
Configure an ENUM
translation rule.
rule tag preference value
match-pattern replacement-rule
domain-name
Not configured by default.
6.
Exit SIP view.
quit
N/A
7.
Enter VoIP entity view.
entity entity-number voip
N/A
8.
Configure ENUM-based SIP
routing by referencing the
ENUM rule group.
address sip enum-group
group-number
Not configured by default.
144
Configuring user information
When accessing the service provider network, configure the user information provided by the service
provider to confirm user identity.
To configure user information:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter POTS voice entity
view.
entity entity-number pots
N/A
Optional.
5.
Configure user information.
uri user-info [ domain
domain-name ]
By default,
number@SIP-device-domain-n
ame or number
@SIP-interface-IP-address is
used to send request messages.
Configuring outbound SIP proxy server
information for a SIP UA
An outbound SIP proxy server resides between an internal network and a public network. An internal
SIP UA sends all request messages to the outbound proxy server to communicate with the registrar
and proxy server on the public network. The outbound proxy server routes the SIP request
messages to the public network according to their Request-Line header fields.
To configure the outbound proxy server information for a SIP UA:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure the outbound
proxy server information for
the SIP UA.
outbound-proxy { dns
domain-name | ipv4 ip-address }
[ port port-number ]
By default, no outbound proxy
server information is configured
for the SIP UA.
Configuring transport layer protocol for SIP calls
This section describes how to configure UDP and TCP as the transport layer protocol for SIP
outgoing and incoming calls.
Configuring UDP or TCP for outgoing SIP calls
The execution of the transport command in SIP client view specifies the global transport layer
protocol for outgoing SIP calls. If you want to configure a different transport layer protocol for
145
individual calls, you can specify the transport layer protocol in corresponding VoIP voice entity view.
If the transport layer protocol configured in VoIP voice entity view and that configured in SIP client
view are different, the former is adopted.
Configuration prerequisites
If a SIP server is used to forward calls, you need to complete the configurations on the SIP server.
Configuration procedure
To specify UDP or TCP as the global transport layer protocol for outgoing SIP calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Specify UDP or TCP as the
global transport layer
protocol for outgoing SIP
calls.
transport { tcp | udp }
By default, UDP is adopted.
To specify UDP or TCP as the transport layer protocol for outgoing SIP calls on a VoIP voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
By default, the global transport
layer protocol is UDP, and no
transport layer protocol is
specified for a VoIP voice entity.
5.
Specify UDP or TCP as the
transport layer protocol for
outgoing SIP calls.
transport { tcp | udp }
If the transport layer protocol is
not specified for a VoIP voice
entity, the global setting is
applied.
This command is effective only
when the type of the VoIP voice
entity is SIP.
NOTE:
The transport layer protocol configured on two communication parties must be the same. That is, if
you execute the transport tcp command on the sender device, you need to execute the listen
transport tcp command on the receiver device.
To set the aging time for TCP connections:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
146
Step
Command
4.
timer connection age tcp
tcp-age-time
Set the aging time for TCP
connections.
Remarks
Optional.
By default, the aging time for TCP
connections is 5 minutes.
Configuring UDP or TCP for incoming SIP calls
Configuration prerequisites
If a SIP server is used to forward calls, you need to complete the configuration at the SIP server end.
Configuration procedure
To specify UDP or TCP as the transport layer protocol for incoming SIP calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Specify UDP or TCP as the
transport layer protocol for
incoming SIP calls.
listen transport { tcp | udp }
Optional.
By default, both the UDP and TCP
listening ports are enabled.
Configuring SIP security
This section describes how to configure TLS for outgoing SIP calls and media flow protocols for SIP
calls.
Configuring TLS for SIP sessions
Configuration prerequisites
•
If a SIP server is used to forward calls, you need to complete the configurations on the SIP
server.
•
Configure an SSL policy on the device. To make sure that the certificate on the device can be
used, be sure that the device system time falls within the validity time of the certificate.
Configuration procedure
To reference the SSL policy:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Reference the SSL policy for
the client.
crypto ssl-client-policy
client-policy-name
Not configured by default.
5.
Reference the SSL policy for
the server.
crypto ssl-server-policy
server-policy-name
147
Not configured by default.
For more information about SSL
Step
Command
Remarks
policies, see Security
Configuration Guide.
To specify TLS as the global transport layer protocol for outgoing SIP calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Specify TLS as the global
transport layer protocol for
outgoing SIP calls.
transport tls
By default, UDP is adopted.
To specify TLS as the transport layer protocol for outgoing SIP calls on a VoIP voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
By default, the global transport
layer protocol is UDP, and no
transport layer protocol is
specified for a VoIP voice entity.
5.
Specify TLS as the transport
layer protocol for outgoing
SIP calls.
If the transport layer protocol is
not specified for a VoIP voice
entity, the global setting is
applied.
transport tls
This command is effective only
when the type of the VoIP voice
entity is SIP.
If you execute this command, the
default port number for the VoIP
voice entity is 5061. For more
information, see the address sip
command in Voice Command
Reference.
NOTE:
The transport layer protocol configured on two communication parties must be the same. That is, if
you execute the transport tls command on the sender device, you need to execute the listen
transport tls command on the receiver device.
To specify TLS as the transport layer protocol for incoming SIP calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
148
Step
Command
Remarks
3.
Enter SIP client view.
sip
N/A
4.
Specify TLS as the transport
layer protocol for incoming
SIP calls.
listen transport tls
By default, the TLS listening port
is disabled.
To set the aging time for TLS connection:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Set the aging time for TLS
connection.
timer connection age tls
tls-age-time
Optional.
By default, the aging time for TLS
connection is 30 minutes.
Configuring media flow protocols for SIP calls
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuration procedure
To specify the media flow protocols for SIP calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
Optional.
By default, RTP is adopted.
4.
Specify the media flow
protocols for SIP calls.
media-protocol { rtp | srtp } *
After you have specified both the
RTP and SRTP protocols as the
media flow protocols for SIP calls,
if the device is the call initiator,
both media flow protocols are
carried in the INVITE message for
the receiver to select. If the device
is the call receiver, the SRTP
protocol is applied for media flow
negotiation. If the negotiation fails,
the RTP protocol is applied.
Specifying the URL scheme for outgoing SIP calls
The device provides two URL schemes: SIP and SIP secure (SIPS). You can choose either as
needed. To ensure transmission security, specify the SIPS scheme.
The execution of the url command in SIP client view specifies the global URL scheme. If you want to
configure a different URL scheme for individual outgoing SIP calls, specify the URL scheme in
149
corresponding VoIP voice entity view. When the URL scheme configured in VoIP voice entity view
and that configured in SIP client view are different, the former is adopted. That is, the VoIP voice
entity configuration takes precedence over global configuration.
Configuration prerequisites
•
If a SIP server is used to forward calls, complete the configuration at the SIP server end.
•
As the SIPS scheme must use TLS, configure TLS as the transport layer protocol of SIP before
specifying the SIPS scheme.
Configuration procedure
To specify the global URL scheme for outgoing SIP calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
Sip
N/A
Optional.
4.
Specify the global URL
scheme for outgoing SIP
calls.
url { sip | sips }
By default, the SIP scheme is
adopted.
You can specify the SIPS scheme
only when the transport layer
protocol is TLS.
To specify the URL scheme for outgoing SIP calls on a VoIP voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
Optional.
By default, no URL scheme is
configured.
5.
Specify the URL scheme for
outgoing SIP calls.
url { sip | sips }
This command is effective only
when the type of the VoIP voice
entity is SIP.
You can specify the SIPS scheme
only when the transport layer
protocol is TLS.
Configuring SIP extensions
Strict SIP routing
Strict SIP routing is supported. In a complicated network environment where a request from SIP UAC
to SIP UAS needs to pass through multiple proxy servers, SIP uses the Route header field and the
Record-Route header field to make sure that requests in the dialog can be routed through these
proxy servers. The strict SIP routing is enabled by default, and you do not need to configure it.
150
Configuring out-of-band SIP DTMF transmission mode
During the communication between caller and callee, the DTMF digits can be transmitted
transparently between them in two ways: in-band and out-of-band. In the in-band approach, the
DTMF digits are encoded in RTP voice packets, and in the out-of-band approach, in SIP messages.
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server. To implement
bidirectional DTMF out-of-band transmission, configure it in the VoIP entity on the calling UA and in
the corresponding POTS voice entity on the called UA. For specific configurations, see "Configuring
out-of-band SIP DTMF transmission mode."
Configuration procedure
To configure the out-of-band SIP DTMF transmission mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots |
voip }
N/A
5.
Configure the out-of-band
SIP DTMF transmission
mode.
outband sip
Inband DTMF transmission mode
is applied by default.
Configuring source IP address binding for SIP messages
Perform this task to specify a source IP address for SIP signaling or media messages by using one of
the following methods.
•
Static IP address binding—Uses a static address as the source IP address.
•
Interface binding—Uses the IP address of a source interface as the source IP address. In a
large-scale network, an interface obtains its IP address from a DHCP or PPPoE server. In this
scenario, you can use this method to configure an interface as the source of SIP signaling and
media flows to avoid manual configuration.
Source IP address binding is supported on Layer 3 Ethernet interfaces, Gigabit Ethernet interfaces,
and dialer interfaces.
For more information about DHCP, see Layer 3—IP Services Configuration Guide.
For more information about PPPoE, see Layer 2—WAN Configuration Guide.
Configuration prerequisites
If a SIP server is used to forward calls, you need to complete the configurations on the SIP server.
Configuration procedure
To configure SIP support for source IP address binding:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
151
Step
Command
Remarks
3.
Enter SIP client view.
sip
N/A
4.
Specify a source IP address
for sent SIP signaling or
media flows.
source-bind { media | signal }
{ interface-type interface-number |
ipv4 ip-address }
By default, no source IP address
is bound for SIP signaling and
media flows.
The following table describes how source address binding works in different conditions:
Condition
Result
•
Configure a source address binding when
ongoing calls exist.
•
A new source address binding for media does not take
effect for ongoing SIP media sessions but takes effect for
subsequent SIP media sessions.
A new source address binding for signal takes effect
immediately for all SIP signaling sessions.
The bound source interface or the
interface whose IP address is set as the
source address is shut down.
The source IP address binding becomes invalid and will not
work until the interface is up. During the shutdown period, the
gateway automatically gets a source IP address for sent
signaling or media flows.
The bound static IP address is removed or
modified, or the bound interface is
removed.
The source IP address binding is removed.
The bound interface is disconnected.
The source IP address binding is cancelled, and restored when
the interface is connected.
Configure a source address binding when
the physical layer or link layer state of the
corresponding interface is down.
The source address binding does not take effect and the
gateway automatically gets a source IP address for packets.
The DHCP lease duration expires and the
bound interface dynamically obtains a new
IP address from the DHCP server
The new IP address will be used as the source IP address.
The SIP registrar is enabled.
The subsequent registration update messages use the source
IP address newly bound for signaling streams to initiate
registration.
Configuring a domain name for the SIP UA
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuration procedure
Configure a domain name for a SIP UA with the sip-domain command. After the configuration, the
domain name address will be carried in the From header field.
To configure a domain name for the SIP UA:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure a domain
name for the SIP UA.
sip-domain domain-name
No domain name is configured for the
SIP UA by default.
152
Configuring SIP compatibility
The devices of some vendors do not strictly follow the SIP protocol. To interoperate with such
devices, configure the SIP compatibility options.
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuring how to obtain the destination number and address
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure the device to
obtain the destination
number from the To header
field for sending a SIP
request.
sip-comp callee
By default, the destination number
is obtained from the request-line,
which is the start line in an SIP
request message.
sip-comp from
By default, the From header field
contains the source address, and
the To header field contains the
destination address.
5.
Configure the device to use
the address in the To header
field as the address in the
From header field.
Enabling the number substitution function
Devices from some vendors cannot recognize the pound sign (#) in call numbers.
To communicate with such devices, enable the number substitution function so that the device
replaces the pound sign in called numbers with the ASCII code %23 for outgoing calls, and replaces
the ASCII code %23 in called numbers with a pound sign for incoming calls. Moreover, the device
replaces the pound sign in the called number with the ASCII code %23 in the Contact header field of
any subsequent request and response.
Typically, the translated called number is limited in length. If a translated called number contains
more than 31 digits, only the first 31 digits are sent. With this length limitation, the device might not
replace all pound signs in a called number. For example, the number
123456789012345678901234567##89 is translated to 123456789012345678901234567%23.
To enable the number substitution function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Enable the number
substitution function.
sip-comp substitute
By default, the number
substitution function is disabled.
Configuring SIP compatibility for fax pass-through and modem pass-through
When the device communicates with devices of other vendors, you can set the following options to
meet your network requirements. After the x-param compatibility option is configured:
•
If the device receives a re-INVITE request with the a=X-modem field, it will reply with a 200 OK
response carrying the a=X-modem field in the SDP field.
153
•
If the device receives a re-INVITE request with the a=X-fax field, it will reply with a 200 OK
response carrying the a=X-fax field.
•
When the device initiates a fax pass-through operation, the a=X-fax field is carried in the
re-INVITE request. When the device initiates a modem pass-through operation, the
a=X-modem field is carried in the re-INVITE request.
After the T.38 compatibility option is configured, the device can recognize T.38-specific description
fields, and fax parameters T38FaxTranscodingJBIG, T38FaxTranscodingMMR, and
T38FaxFillBitRemoval. These are in the SDP fields of the re-INVITE requests and 200 OK
responses, and do not contain :0.
By default, the compatibility options are not carried in re-INVITE requests. After the compatibility
option command is configured, the compatibility options will be carried in re-INVITE requests.
To configure SIP compatibility for fax pass-through and modem pass-through:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure SIP compatibility
for fax pass-through and
modem pass-through.
sip-comp { t38 | x-parameter } *
By default, the compatibility
options are not carried in
re-INVITE requests.
Configuring the parameter for communicating with a VCX device
A VCX device manages various licenses corresponding to different types of telephones, such as
basic phone, and business phone. Phones of different types use different licenses.
To enable the VCX device to differentiate phone types, configure the Contact header fields of
REGISTER messages sent to the VCX device to contain the dt parameter, which specifies the
device (phone) type.
To configure the parameter for communicating with a VCX device:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure the Contact
header fields of the
REGISTER messages to
contain the dt parameter.
sip-comp dt
By default, the Contact header
fields of the REGISTER
messages do not contain the dt
parameter.
Configuring user-agent and server header fields
The User-Agent header field can contain the product name and product version number of the UAC
in a SIP request message, while the Server header field can contain the product name and product
version number of the UAS in a SIP response message. Revealing the specific software version of
the user agent and server might allow the user agent and server to become more vulnerable to
attacks against software that is known to contain security holes. Therefore, it is stipulated in RFC
3261 that the User-Agent header field and the Server header field should both be a configurable
option.
154
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuration procedure
To configure User-Agent and Server header fields:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Configure the User-Agent
header field in a SIP request.
sip-comp agent
product-name product-version
By default, the User-Agent header
field in a SIP request is not
configured.
5.
Configure the Server header
field in a SIP response.
sip-comp server
product-name product-version
By default, the Server header field in
a SIP response is not configured.
Configuring SIP extensions for caller identity and privacy
When a PSTN user originates a call to another PSTN user over a SIP network, the caller identity
presentation can be restricted at the called side or the caller identity can be transparently transmitted
to the called PSTN user by configuring the SIP extension header fields.
The SIP extensions for caller identity and privacy feature can be implemented by adding the
P-Preferred-Identity, P-Asserted-Identity, or Remote-Party-ID header fields.
•
When the P-Preferred-Identity or P-Asserted-Identity header field is added, the Privacy header
field will be added. When the Privacy header field is set to none, the caller identity presentation
is allowed. When the Privacy header field is set to id, the caller identity presentation is
restricted.
•
Remote-Party-ID header field: privacy=off indicates the caller identity presentation and
privacy=full indicates the caller identity screening. The calling information can be transparently
transmitted by adding the Remote-Party-ID header field.
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuration procedure
To configure the SIP extensions for caller identity and privacy feature:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
4.
Add the P-Asserted-Identity
header field or the
P-Preferred-Identity header
field.
privacy { asserted | preferred }
Add the Remote-Party-ID
header field.
remote-party-id
5.
Optional.
By default, neither the
P-Preferred-Identity header field
nor the P-Asserted-Identity
header field is added.
Optional.
155
By default, the Remote-Party-ID
header field is not added.
NOTE:
The Remote-Party-ID header field can be used together with the P-Preferred-Identity header field or
P-Asserted-Identity header field. If so, the Remote-Party-ID header field takes precedence over the
P-Preferred-Identity header field or the P-Asserted-Identity header field.
Configuring call release cause code mapping
Configuration prerequisites
If a SIP server is used to forward calls, complete the configurations on the SIP server.
Configuration procedure
No matter whether a voice call is cleared normally or abnormally, a message with the call release
cause code will be sent. The default SIP status code to PSTN release cause code mappings and
PSTN release cause code to SIP status mappings are used for communication between a SIP
network and a PSTN. For more information about these default mappings, see Voice Command
Reference. To adapt to more complex network applications, you can change the default mappings
through command lines.
To configure PSTN release cause code to SIP status code mappings and SIP status code to PSTN
release cause code mappings:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
Optional.
4.
5.
Configure a PSTN
release cause code to
SIP status code
mapping.
reason-mapping pstn
pstn-code sip sip-code
Configure a SIP status
code to PSTN release
cause code mapping.
reason-mapping sip sip-code
pstn pstn-code
For the default PSTN release cause
code to SIP status code mappings, see
Voice Command Reference. Because
the PSTN release cause code 16
corresponds to a SIP request message,
instead of a SIP status code, you can
configure no SIP status code for 16.
Optional.
For the default SIP status code to PSTN
release cause code mappings, see
Voice Command Reference.
Configuring periodic refresh of SIP sessions
In a high-volume traffic environment, if a BYE message gets lost for a session, the call proxy server
will not know that the session has ended, and thus still maintains the state information for the call,
which wastes resources of the server. To solve this problem, the RFC 4082 defines a session timer
mechanism for SIP sessions: the UA sends periodic re-INVITE or UPDATE requests (referred to as
session refresh requests) to notify the proxy server about the current state of the session. The
interval for sending session refresh requests is determined through the negotiation of both sides.
Two new header fields are added to the session refresh requests:
•
Session-Expires—Conveys the maximum session expiration time, that is, if no refresh request
is received during this time, the session is considered ended.
156
•
Min-SE—Conveys the minimum session expiration time, which is used to avoid frequent
refresh requests from occupying network bandwidth.
Configuration prerequisites
If calls need to be routed to a SIP server, complete the configuration on the SIP server first.
Configuration procedure
To configure periodic refresh of SIP sessions:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
Optional.
4.
Enable periodic refresh of
SIP sessions and set the
maximum and minimum
session expiration time.
timer session-expires seconds
[ minimum min-seconds ]
By default,.
•
The periodic refresh of SIP
sessions is not enabled
automatically. That is, if
periodic refresh of SIP
sessions is disabled on the
called party but enabled on
the calling party, the called
party will enable periodic
refresh of SIP sessions after
negotiation.
•
The minimum session
duration is 90 seconds.
Enabling early media negotiation
With early media negotiation enabled, if the device is the called party, it sends a 183 session
progress response with media information to the calling party to perform early media negotiation and
automatically plays tones to the calling party. The called party can play only common ring back tones
to the calling party. However, in some actual applications, such as the caller ring back tone (CRBT)
service, servers are required to play the ring back tones to the calling party. Therefore, you need to
disable early media negotiation on the called party for those applications. After that, the called party
sends a 180 ringing response without media information to the calling party, and the calling party
receives only CRBTs played by the server instead of the common ring back tones played by the
called party.
To enable early media negotiation:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
Optional.
4.
Enable early media
negotiation on the device.
early-media enable
157
Enabled by default. In other
words, when the device is the
called party, it sends a 183
session progress response with
media information to the called
party.
Configuring VRF-aware SIP
Before you perform this task, you must have completed L3VPN configurations.
When you configure VRF-aware SIP, follow these restrictions and guidelines:
•
Configure this feature when no SIP services are running. This feature takes effect on all SIP
services such as SIP calling, registration, and subscription.
•
To use SIP source binding, the VPN instance associated with the source interface must be the
same as the VPN instance specified in this task.
To configure VRF-aware SIP:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Specify a VPN instance for
SIP.
vpn-instance vpn-instance-name
Optional.
By default, no VPN instance is
specified for SIP.
Displaying and maintaining SIP UAs
Task
Command
Remarks
Display all call statistics of the SIP UA.
display voice sip
call-statistics [ | { begin |
exclude | include }
regular-expression ]
Available in any view.
Display information about SIP connections
over a specific transport layer protocol.
display voice sip connection
{ tcp | tls } [ | { begin | exclude |
include } regular-expression ]
Available in any view.
Display the registration state of the SIP UA.
display voice sip
register-state [ | { begin |
exclude | include }
regular-expression ]
Available in any view.
Display the PSTN release cause code to SIP
status code mappings or the SIP status code
to PSTN release cause code mappings.
display voice sip
reason-mapping { pstn-sip |
sip-pstn } [ | { begin | exclude |
include } regular-expression ]
Available in any view.
Display ENUM translation rule group
configuration.
display voice enum-group { all
| mark group-number } [ | { begin
| exclude | include }
regular-expression ]
Available in any view.
Display DNS dynamic cache information.
display dns host { naptr | srv }
[ | { begin | exclude | include }
regular-expression ]
Available in any view.
Display SIP DNS records.
display voice sip dns-record [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Display the temporarily saved SIP user
identifications and the mapping information of
display voice sip
dynamic-contact-address [ |
Available in any view.
158
Task
Command
the users’ contact addresses.
{ begin | exclude | include }
regular-expression ]
Remarks
Clear all call statistics of the SIP UA
reset voice sip statistics
Available in user view.
Clear a specified SIP connection over a
specific transport layer protocol.
reset voice sip connection
{ tcp | tls } id conn-id
Available in user view.
Clear the DNS dynamic cache.
reset dns host { naptr | srv }
Available in user view.
Clear SIP DNS lookup records.
reset voice sip dns-record
Available in user view.
SIP UA configuration examples
This section provides SIP UA configuration examples.
Configuring direct calling for SIP UAs
Network requirements
Two routers can directly call each other as SIP UAs.
Figure 42 Network diagram
Configuration procedure
Routing-related configurations are beyond the scope of this example. This example assumes that
Router A and Router B are reachable to each other.
1.
Configure Router A:
# Configure the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
[Router1-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
159
[RouterA-voice-dial-entity1111] match-template 1111
2.
Configure Router B:
# Configure the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial-entity2222] quit
[RouterB-voice-dial]entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
Configuration verification
With the above configuration, you can use telephone 1111 to call telephone 2222, or use telephone
2222 to call telephone 1111.
•
On Router A, the global authentication method is adopted. All numbers use the same
authentication username routerA and the same password 1234.
•
On Router B, the authentication username and password of number 2222 are routerB and
1234 respectively.
Configuring proxy server involved calling for SIP UAs
Network requirements
Two routers work as SIP UAs and SIP calls are made through a SIP server.
Figure 43 Network diagram
Configuration procedure
Routing-related configurations are beyond the scope of this example. This example assumes that
Router A, Router B, and the SIP server are reachable to each other.
The configuration of SIP server is not dealt with in this example because it varies with devices.
1.
Configure Router A:
# Configure the Ethernet interface.
<RouterA> system-view
160
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure SIP.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 192.168.2.3
[RouterA-voice-sip] proxy ipv4 192.168.2.3
[RouterA-voice-sip] user routerA password cipher 1234
[RouterA-voice-sip] register-enable on
[RouterA-voice-sip] quit
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] quit
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip proxy
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
2.
Configure Router B:
# Configure the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Configure SIP.
[RouterB] voice-setup
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 192.168.2.3
[RouterB-voice-sip] proxy ipv4 192.168.2.3
[RouterB-voice-sip] register-enable on
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial-entity2222] user routerB password cipher 1234
[RouterB-voice-dial-entity2222] quit
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip proxy
[RouterB-voice-dial-entity1111] match-template 1111
[RouterB-voice-dial-entity1111] quit
161
Configuration verification
After the local numbers of the two sides are registered on the registrar successfully, you can make
calls between telephone 1111 and telephone 2222 through the proxy server.
Configuring DNS involved calling for SIP UAs
Network requirements
Two routers work as SIP UAs and SIP calls are made through DNS.
Figure 44 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
# Map the IP address 192.168.2.2 to the host name cc.news.com.
<Sysname> system-view
[Sysname] ip host cc.news.com 192.168.2.2
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip dns cc.news.com
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
2.
Configure Router B:
# Configure the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
162
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial-entity2222] quit
[RouterB-voice-dial]entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
Configuration verification
After the above configuration, calls between telephone 1111 and telephone 2222 are made through
the DNS server.
Configuring out-of-band SIP DTMF transmission mode
Network requirements
Two routers work as SIP UAs. The calling and called parties adopt DTMF SIP out-of-band
transmission to make the transmission of DTMF digits more reliable.
Figure 45 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure the IP address of the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
# Configure the voice entity.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] outband sip
[RouterA-voice-dial-entity2222] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] outband sip
2.
Configure Router B:
163
# Configure the IP address of the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
# Configure the voice entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
[RouterB-voice-dial-entity1111] outband sip
[RouterB-voice-dial-entity2222] quit
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial-entity2222] outband sip
Configuration verification
After the call is established, if one side presses the telephone keys, the DTMF digits are transmitted
to the other side using out of band signaling, and the other side hears short DTMF tones from the
handset.
Configuring SIP extensions for caller identity and privacy
Network requirements
Two routers Router A and Router B work as SIP UAs. Telephone 1111 calls telephone 2222. It is
required that the calling number 1111 not be displayed on telephone 2222.
Figure 46 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure the IP address of the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure the voice entities.
[RouterA] voice-setup
[RouterA-voice] dial-program
164
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] quit
[RouterA-voice-dial] quit
[RouterA-voice] quit
# Disable the calling voice subscriber line from sending the calling number to the remote end.
[RouterA] subscriber-line 1/0
[RouterA-subscriber-line1/0] undo cid send
[RouterA-subscriber-line1/0] quit
# Configure the P-Asserted-Identity header field.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] privacy asserted
2.
Configure Router B:
# Configure the IP address of the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Configure the voice entity.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial] entity 1111 voip
[RouterA-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterA-voice-dial-entity1111] match-template 1111
Configuration verification
When telephone 1111 calls telephone 2222, the calling number 1111 will not be displayed on
telephone 2222.
Configuring TCP to carry outgoing SIP calls
Network requirements
Two routers Router A and Router B work as SIP UAs. It is required that the SIP calls from Telephone
1111 to telephone 2222 be carried over TCP.
165
Figure 47 Network diagram
Confguration procedure
1.
Configure Router A:
# Configure the IP address of the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Specify TCP as the global transport layer protocol for outgoing SIP calls.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] transport tcp
[RouterA-voice-sip] quit
# Configure the voice entities.
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] quit
[RouterA-voice-dial] quit
2.
Configure Router B:
# Configure the IP address of the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Specify TCP as the transport layer protocol for incoming SIP calls. (Optional, because the
TCP listening port is enabled by default.)
[RouterB] voice-setup
[RouterB-voice] sip
[RouterB-voice-sip] listen transport tcp
[RouterB-voice-sip] quit
# Configure the voice entities.
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
166
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
Configuration verification
SIP calls from Telephone 1111 to telephone 2222 are carried over TCP. You can view the information
of all TCP connections using the display voice sip connection tcp command.
Configuring TLS to carry outgoing SIP calls
Network requirements
Two routers Router A and Router B work as SIP UAs. It is required that the SIP calls between the two
parties be carried over TLS.
Figure 48 Network diagram
Confguration procedure
The certification authority (CA) server runs RSA Keon in this configuration example.
For information about how to configure the TLS policy, see Security Configuration Guide.
IMPORTANT:
To make sure the certificate on the device can be used, be sure that the device system time falls
within the validity time of the certificate.
1.
Configure Router A:
# Configure the IP address of the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Create a PKI entity aaa, enter its view, and then configure the common name of the entity as
RouterA.
[RouterA] pki entity aaa
[RouterA-pki-entity-aaa] common-name RouterA
[RouterA-pki-entity-aaa] quit
# Create a PKI domain voice, enter its view, and then specify the trusted CA as voice.
[RouterA] pki domain voice
[RouterA-pki-domain-voice] ca identifier voice
# Specify the URL of the registrar in the format of http://host:port/Issuing Jurisdiction ID,
where Issuing Jurisdiction ID is a hexadecimal character string generated on the CA server.
167
Then, specify the authority for certificate request as CA, and the entity for certificate request as
aaa.
[RouterA-pki-domain-voice] certificate request url
http://192.168.0.88:446/bd0683e5a369eb4edbb4ef502eaca6ec42d24e97
[RouterA-pki-domain-voice] certificate request from ca
[RouterA-pki-domain-voice] certificate request entity aaa
[RouterA-pki-domain-voice] quit
# Create local RSA key pairs.
[RouterA] public-key local create rsa
# Retrieve the CA certificate from the certificate issuing server.
[RouterA] pki retrieval-certificate ca domain voice
# Request a local certificate from the CA.
[RouterA] pki request-certificate domain voice
# Create an SSL server policy named server and configure the policy to use PKI domain voice.
[RouterA] ssl server-policy server
[RouterA-ssl-server-policy-server] pki-domain voice
# Create an SSL client policy named client and configure the policy to use PKI domain voice.
[RouterA] ssl client-policy client
[RouterA-ssl-client-policy-server] pki-domain voice
# Reference the created SSL server and client policies for SIP, and then specify TLS as the
transport layer protocol for both outgoing and incoming SIP calls.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] crypto ssl-server-policy server
[RouterA-voice-sip] crypto ssl-client-policy client
[RouterA-voice-sip] listen transport tls
[RouterA-voice-sip] transport tls
[RouterA-voice-sip] quit
# Configure the voice entities.
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2 port 5061
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] quit
[RouterA-voice-dial] quit
2.
Configure Router B:
# Configure the IP address of the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Create a PKI entity aaa, enter its view, and then configure the common name of the entity as
RouterB.
[RouterB] pki entity aaa
168
[RouterB-pki-entity-aaa] common-name RouterB
[RouterB-pki-entity-aaa] quit
# Create a PKI domain voice, enter its view, and then specify the trusted CA as voice.
[RouterB] pki domain voice
[RouterB-pki-domain-voice] ca identifier voice
# Specify the URL of the registrar for certificate request. The URL is in the format of
http://host:port/Issuing Jurisdiction ID, where Issuing Jurisdiction ID is a hexadecimal
character string generated on the CA server. Then, specify the authority for certificate request
as CA, and specify the entity for certificate request as h3c.
[RouterB-pki-domain-voice] certificate request url
http://192.168.0.88:446/bd0683e5a369eb4edbb4ef502eaca6ec42d24e97
[RouterB-pki-domain-voice] certificate request from ca
[RouterB-pki-domain-voice] certificate request entity aaa
[RouterB-pki-domain-voice] quit
# Create local RSA key pairs.
[RouterB] public-key local create rsa
# Retrieve the CA certificate from the certificate issuing server.
[RouterB] pki retrieval-certificate ca domain voice
# Request a local certificate from the CA.
[RouterB] pki request-certificate domain voice
# Create an SSL server policy named server and configure the policy to use PKI domain voice.
[RouterB] ssl server-policy server
[RouterB-ssl-server-policy-server] pki-domain voice
# Create an SSL client policy named client and configure the policy to use PKI domain voice.
[RouterB] ssl client-policy client
[RouterB-ssl-client-policy-server] pki-domain voice
# Reference the created SSL server and client policies for SIP, and then specify TLS as the
transport layer protocol for both outgoing and incoming SIP calls.
[RouterB] voice-setup
[RouterB-voice] sip
[RouterB-voice-sip] crypto ssl-server-policy server
[RouterB-voice-sip] crypto ssl-client-policy client
[RouterB-voice-sip] listen transport tls
[RouterB-voice-sip] transport tls
[RouterB-voice-sip] quit
# Configure the voice entities.
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 port 5061
[RouterB-voice-dial-entity1111] match-template 1111
Configuration verification
SIP calls from Telephone 1111 to telephone 2222 are carried over TLS. You can view the information
of all TLS connections using the display voice sip connection tls command.
169
Configuring SIPS URL scheme for outgoing SIP calls
Network requriements
Two routers Router A and Router B work as SIP UAs. It is required that the SIP calls from Telephone
1111 to telephone 2222 use the SIPS URL scheme.
Figure 49 Network diagram
Configuration procedure
You can use the SIPS URL scheme only when the transport layer protocol is TLS. Therefore, before
performing following configurations, you need to configure TLS for SIP as described in "Configuring
TLS to carry outgoing SIP calls."
When a call is initiated, the TLS listening port of the calling party must be open; otherwise, the call
connection cannot be established.
For information about how to configure the TLS policy, see Security Configuration Guide.
1.
Configure Router A:
# Configure the IP address of the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Reference the created SSL server and client policies for SIP. Specify TLS as the transport
layer protocol for both outgoing and incoming SIP calls; and specify URL scheme for the
outgoing SIP call as SIPS.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] crypto ssl-server-policy server
[RouterA-voice-sip] crypto ssl-client-policy client
[RouterA-voice-sip] listen transport tls
[RouterA-voice-sip] url sips
[RouterA-voice-sip] transport tls
[RouterA-voice-sip] quit
# Configure voice entities.
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2 port 5061
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
170
[RouterA-voice-dial-entity1111] quit
[RouterA-voice-dial] quit
2.
Configure Router B:
# Configure the IP address of the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Reference the created SSL server and client policies for SIP. Specify TLS as the transport
layer protocol for both outgoing and incoming SIP calls, and specify URL scheme for the
outgoing SIP call as SIPS.
[RouterB] voice-setup
[RouterB-voice] sip
[RouterB-voice-sip] crypto ssl-server-policy server
[RouterB-voice-sip] crypto ssl-client-policy client
[RouterB-voice-sip] listen transport tls
[RouterB-voice-sip] url sips
[RouterB-voice-sip] transport tls
[RouterB-voice-sip] quit
# Configure voice entities.
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 port 5061
[RouterB-voice-dial-entity1111] match-template 111
Configuration verification
SIP calls between the two parties use the SIPS URL scheme.
Configuring SRTP for SIP calls
Network requriements
Two routers Router A and Router B work as SIP UAs. It is required that SIP calls use the SRTP
protocol to protect call conversations.
Figure 50 Network diagram
Configuration procedure
1.
Configure Router A:
171
# Configure the IP address of the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Specify SRTP as the media flow protocol for SIP calls.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] media-protocol srtp
[RouterA-voice-sip] quit
# Specify 1111 as a local number of POTS voice entity 1111.
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
# Configure VoIP voice entity 2222, and configure the IP address of the peer VoIP gateway as
192.168.2.2, and the called number as 2222.
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2 port 5060
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
2.
Configure Router B:
# Configure the IP address of the Ethernet interface.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Specify SRTP as the media flow protocol for SIP calls.
[RouterB] voice-setup
[RouterB-voice] sip
[RouterB-voice-sip] media-protocol srtp
[RouterB-voice-sip] quit
# Specify 2222 as a local number of POTS voice entity 2222.
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
# Configure VoIP voice entity 1111, and configure the IP address of the peer VoIP gateway as
192.168.2.1, and the called number as 1111.
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 port 5060
[RouterB-voice-dial-entity1111] match-template 1111
Configuration verification
SIP calls use the SRTP protocol to encrypt and authenticate media flows, protecting call
conversations.
172
Troubleshooting
SIP UA configurations are simple. You can identify most problems by viewing configuration and
debugging information.
Failed to set up calls in the proxy server approach to SIP
routing
Symptom
The UA could not set up calls when the proxy server approach was adopted to SIP routing.
Solution
Do the following:
•
Execute the display current-configuration command to check for information about the SIP
proxy server or the registrar.
•
If the proxy server or the registrar is not configured, configure it in SIP client view.
Failed to register with the registrar
Symptom
The UA could not register with the registrar.
Solution
Check that:
•
A route to the registrar is available.
•
The local configuration is consistent with that of the registrar.
•
A SIP authentication password is configured with the user command on the voice gateway if
authentication is necessary.
•
The configuration of realm in the authentication information is consistent with that on the
registrar. If the realm argument is locally configured, make sure the local configuration is the
same as that on the registrar. Otherwise, the voice gateway will not initiate an authentication
request. If the realm argument is not locally configured, the voice gateway will not judge the
configuration of realm on the server and consider the server is trusted.
Failed to set up point-to-point calls
Symptom
The UA could not set up point-to-point calls.
Solution
Check that the IP address and the port number of the remote voice gateway are correctly configured.
Failed to send register requests
Symptom
The UA does not send REGISTER messages.
173
Solution
Do the following:
•
Execute the debugging voice sip command to see whether REGISTER messages are being
sent.
•
If no REGISTER messages are sent, enable the registration function with the register-enable
command.
Failed to set up point-to-point SIP calls over TLS
Symptom
The UA could not set up point-to-point SIP calls over TLS when the basic voice gateway settings are
correct.
Solution
Check that:
•
The devices of both sides have certificates.
•
TLS policies have been configured.
•
TLS policies have been referenced in SIP client view.
•
TLS is specified as the transport layer protocol for outgoing SIP calls. (You can specify the
transport layer protocol in either VoIP voice entity view or SIP client view.)
•
Listening ports are enabled on both the calling device and the called device.
•
The number of the port corresponding to the destination address is set to 5061 for VoIP voice
entity.
174
Configuring SIP local survival
IP phones have been deployed throughout the headquarters and branches of many enterprises and
organizations. Typically, a voice server is deployed at the headquarters to control calls originated by
IP phones at the branches.
The local survival feature enables the voice router at a branch to automatically detect the reachability
to the headquarter voice server, and process calls originated by attached IP phones when the
headquarters voice server is unreachable. The headquarters voice router will take over call services
from the branch voice router when the failure is removed.
Figure 51 shows a typical network diagram for the local survival feature.
Figure 51 Network diagram
Branch A
WAN
Server
Headquarters
Branch B
PSTN
Branch C
The following describes the local survival feature in detail:
•
When a WAN link from a branch to the headquarters is normal, all IP phones at the branch are
registered with the headquarters voice server and the headquarters voice server processes
calls originated by branch IP phones.
•
When the WAN link to the headquarters or the primary server fails:
{
{
{
•
The branch voice router can accept registrations from its attached IP phones.
The branch voice router ensures the normal call services between its IP phones, between
its IP phones and FXS interfaces, and between its FXS interfaces.
IP phone users at the branch can place or receive PSTN calls through FXS interfaces on the
voice router.
When the WAN link or the primary server recovers, the branch voice router rejects registrations
from IP phones and the headquarters voice server takes over call processing.
Hardware compatibility with SIP local survival
SIP local survival is not available on the following routers:
•
MSR800.
•
MSR 900.
175
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
Configuration task list
Task
Remarks
Configuring an operation mode for the local SIP server
Required.
Configuring user information
Optional.
Specifying a trusted node
Optional.
Configuring call authority control
Optional.
Optional.
Configuring an area prefix
Applicable to calls initiated from external users to
internal users.
Optional.
Configuring a call route
Applicable to calls initiated from internal users to
external users.
Configuring an operation mode for the local SIP
server
The local SIP server can operate in either of the following modes:
•
Alone mode—The local SIP server operating in the alone mode act as a small voice server.
The probe remote-server ipv4 command is invalid in the alone mode.
•
Alive mode—When operating in the alive mode, the local SIP server supports the survival
feature between itself and the remote server. That is, when communication with the remote
server fails, the local SIP server accepts registrations and calls. When communication resumes,
the remote server accepts registrations and calls again, and the local SIP server rejects
registrations and calls. In alive mode, Options messages will periodically be sent to the remote
server if the probe remote-server ipv4 command is configured.
To configure the alone mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Configure the local
SIP server to operate
in the alone mode.
mode alone-server
176
By default, the local SIP server operates
in the alone mode.
Note that you can change the operation
mode of the local SIP server only when
Step
Command
Remarks
the local SIP server is disabled.
By default, no IP address is configured,
that is, there is no local SIP server.
5.
Configure the IP
address of the local
SIP server as the IP
address of a local
interface.
server-bind ipv4 ipv4-address
[ port port-number ] [ expires
time-interval ]
Note that the ipv4-address argument can
be the IP address of an interface on the
local router, or a loopback address such
as 127.0.0.1. Because the local SIP
server cannot accept registrations from
users when the server IP address is set
to 127.0.0.1, the IP address of an
interface on the local router is
recommended.
You can configure this command only
when the local SIP server is disabled.
6.
Enable the local SIP
server.
By default, the local SIP server is
disabled.
server enable
The functions of the local SIP server can
take effect after you execute this
command.
To configure the alive mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Configure the local
SIP server to operate
in the alive mode.
By default, the local SIP server operates
in the alone mode.
mode alive-server
Note that you can change the operation
mode of the local SIP server only when
the local SIP server is disabled.
By default, no IP address is bound, that
is, there is no local SIP server.
5.
Set the local SIP
server address to the
IP address of an
interface on the local
router.
server-bind ipv4 ipv4-address
[ port port-number ] [ expires
time-interval ]
Note that the ipv4-address argument can
be the IP address of an interface on the
local router, or a loopback address such
as 127.0.0.1. Because the local SIP
server cannot accept registrations from
users when the server IP address is set
to 127.0.0.1, the IP address of an
interface on the local router is
recommended.
You can configure this command only
when the local SIP server is disabled.
Optional.
6.
7.
Configure the
keepalive probe.
probe remote-server ipv4
ipv4-address [ port port-number ]
[ keepalive time-interval ]
Enable the local SIP
server.
server enable
By default, the keepalive probe is not
configured.
This command is valid only when the
local SIP server operates in the alive
mode, and can be configured only when
the local SIP server is disabled.
By default, the local SIP server is
disabled.
177
Step
Command
Remarks
The functions of the local SIP server can
take effect after you execute this
command.
Configuring user information
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Create a user to be
registered with and enter
register user view.
register-user tag
By default, no user is created to
be registered.
Configure a DN for the user.
number party-number
By default, no DN is configured for
the user.
5.
Optional.
By default, no authentication
information is configured for the
user.
6.
Configure authentication
information for the user.
authentication username
username password { cipher |
simple } password
When accepting registrations and
calls from the user, the local SIP
server needs to check the user’s
authentication information for
validity. The user can register with
the local SIP server to originate
calls only after passing
authentication.
Optional.
7.
Configure the maximum
registration interval.
expires time-interval
By default, the maximum
registration interval is configured
with the server-bind ipv4
command.
Specifying a trusted node
A trusted node can directly originate calls without being authenticated by the local SIP server. You do
not need to configure user information for the number of the trusted node on the local SIP server.
To specify a trusted node:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Specify a trusted node.
trusted-point ipv4 ipv4-address
[ port port-number ]
By default, no trusted node is
specified.
178
Configuring call authority control
This section describes how to configure call authority control.
Configuring a call rule
The local SIP server supports the call authority control feature. Define different call rules and apply
them in different views to control the call authorities of users within the jurisdiction.
To configure a call rule:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Enter call rule set view.
call-rule-set
N/A
5.
Create a call rule and enter
call rule view.
service tag
N/A
Configure a call rule.
rule tag { deny | permit }
{ incoming | outgoing } { pattern |
any }
6.
By default, no call rule is
configured.
You can configure up to 32 call
rules.
Applying a call rule set
You can use the srs command to apply a call rule set in SIP server view or register user view.
When different call rule sets are applied in SIP server view, register user view of the calling user, and
register user view of the called user, the local SIP server first processes the call rule set applied in
SIP server view, then the one applied in register user view of the calling user, and finally the one
applied in register user view of the called user.
To apply a call rule set:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Apply a call rule set in SIP
server view.
srs tag
By default, no call rule set is applied
in SIP server view.
To apply a call rule set in register user view:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Create a user to be
register-user tag
By default, no user is created to be
179
Step
Command
registered with and enter
register user view.
5.
Apply a call rule set in
register user view.
Remarks
registered.
By default, no call rule set is applied
register user view.
srs tag
Only one call rule set can be applied
in register user view.
Configuring an area prefix
When the local SIP server is connected to the public network, external users can originate calls to
internal users registered with the local SIP server. For calls from external users to internal users, the
local SIP server removes the configured area prefix from each called number to converts it to an
internal short number. For example, if an external user dials number 01050009999, the local SIP
server checks whether any area prefix matches the called number. If the area prefix 0105000 is
available, the local SIP server removes the prefix 0105000 from the called number and sends the call
to 9999.
To configure an area prefix:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
By default, no area prefix is
configured.
4.
Configure an area prefix.
area-prefix prefix
If multiple area prefixes are
configured, the local SIP server
adopts longest match to deal with
a called number.
Configuring a call route
The local SIP server uses a static routing table to forward outgoing calls. If the called number
matches a static route, the local SIP server forwards the call to the specified destination. The called
number does not need to register on the local SIP server. For example, as an external number,
5552000 does not need to register on the local SIP server. Configure a static route entry with the
area prefix of 333 and called number of 5552000 on the local SIP server. Upon receiving a call from
local number 1000 to external number 5552000, the local SIP server adds the area prefix 333 to the
calling number, and forwards the call to the destination specified in the static route entry.
To configure a call route entry:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP server view.
sip-server
N/A
4.
Enter call route view.
call-route
N/A
5.
Configure a call route.
trunk tag called-number called-pattern
By default, no call route is
180
Step
Command
Remarks
ipv4 dest-ip-addr [ port port-number ]
[ area-prefix prefix ]
configured.
Displaying and maintaining the SIP local survival
feature configuration
Task
Command
Display information of a registered
user or all registered users.
display voice sip-server
register-user { tag | all } [ |
{ begin | exclude | include }
regular-expression ]
Display the server resource
information.
display voice sip-server
resource-statistic [ | { begin |
exclude | include }
regular-expression ]
Remarks
Available in any view.
SIP local survival feature configuration examples
This section provides configuration examples for SIP local survival.
Configuring the local SIP server to operate in the alone mode
Network requirements
Configure the local SIP server on Router C to operate in alone mode so that the phones register with,
and can make and receive calls through, the local SIP server. See Figure 52.
Figure 52 Network diagram
Configuration procedure
1.
Configure Router C:
# Configure the router to operate in the alone mode.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] sip-server
[RouterC-voice-server] server-bind ipv4 2.1.1.2
[RouterC-voice-server] server enable
# Configure authentication information for Phone 1000 and Phone 5000.
[RouterC-voice-server] register-user 1000
[RouterC-voice-server-user1000] number 1000
[RouterC-voice-server-user1000] authentication username 1000 password simple 1000
[RouterC-voice-server-user1000] quit
181
[RouterC-voice-server] register-user 5000
[RouterC-voice-server-user5000] number 5000
[RouterC-voice-server-user5000] authentication username 5000 password simple 5000
2.
Configure Router A:
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 2/0
[RouterA-voice-dial-entity1000] user 1000 password simple 1000
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] entity 5000 voip
[RouterA-voice-dial-entity5000] address sip proxy
[RouterA-voice-dial-entity5000] match-template 5000
[RouterA-voice-dial-entity5000] quit
[RouterA-voice-dial] quit
# Enable SIP registration.
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 2.1.1.2
[RouterA-voice-sip] register-enable on
3.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 5000 pots
[RouterB-voice-dial-entity5000] match-template 5000
[RouterB-voice-dial-entity5000] line 2/0
[RouterB-voice-dial-entity5000] user 5000 password simple 5000
[RouterB-voice-dial-entity5000] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip proxy
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice] quit
# Enable SIP registration.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 2.1.1.2
[RouterB-voice-sip] register-enable on
4.
Verify the configurations:
[RouterC-voice-server-user5000] display voice sip-server register-user all
user
number
status
address
----------------------------------------------------------------------1000
1000
online
1.1.1.1:5060
5000
5000
online
182
2.1.1.1:5060
Phone 1000 and Phone 5000 are successfully registered with the local SIP server Router C and
they can communicate with each other.
Configuring the local SIP server to operate in the alive mode
(method 1)
Network requirements
Router A and Router B carry out call services through VCX. To ensure the normal call services on
Router A and Router B when VCX fails, configure the local SIP server on Router C to operate in alive
mode, so that calls can be originated or received through Router C. See Figure 53.
Figure 53 Network diagram
Configuration procedure
1.
Configure Router C:
# Configure the router to operate in the alive mode.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] sip-server
[RouterC-voice-server] server-bind ipv4 2.1.1.2
[RouterC-voice-server] mode alive-server
[RouterC-voice-server] probe remote-server ipv4 3.1.1.1
[RouterC-voice-server] server enable
# Configure number registration information for the phones.
[RouterC-voice-server] register-user 1000
[RouterC-voice-server-user1000] number 1000
[RouterC-voice-server-user1000] authentication username 1000 password simple 1000
[RouterC-voice-server-user1000] quit
[RouterC-voice-server] register-user 5000
[RouterC-voice-server-user5000] number 5000
[RouterC-voice-server-user5000] authentication username 5000 password simple 5000
2.
Configure Router A:
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
183
[RouterA-voice-dial-entity1000] line 2/0
[RouterA-voice-dial-entity1000] user 1000 password simple 1000
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] entity 5000 voip
[RouterA-voice-dial-entity5000] address sip proxy
[RouterA-voice-dial-entity5000] match-template 5000
[RouterA-voice-dial-entity5000] quit
[RouterA-voice-dial] quit
# Configure and enable SIP registration.
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 3.1.1.1
[RouterA-voice-sip] registrar ipv4 2.1.1.2 slave
[RouterA-voice-sip] register-enable on
3.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 5000 pots
[RouterB-voice-dial-entity5000] match-template 5000
[RouterB-voice-dial-entity5000] line 2/0
[RouterB-voice-dial-entity5000] user 5000 password simple 5000
[RouterB-voice-dial-entity5000] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip proxy
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] quit
# Configure and enable SIP registration.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 3.1.1.1
[RouterB-voice-sip] registrar ipv4 2.1.1.2 slave
[RouterB-voice-sip] register-enable on
4.
Verify the configurations:
[RouterC-voice-server-user5000] display voice sip-server register-user all
user
number
status
address
---------------------------------------------------------------------1000
1000
online
1.1.1.1:5060
5000
5000
online
2.1.1.1:5060
Phone 1000 and Phone 5000 have already registered with the local SIP server, Router C, which
periodically sends Options messages to VCX to detect the link reachability. If the link is
unreachable, the local SIP server accepts registrations and calls originated by Phone 1000 and
Phone 5000. If the link is reachable, the local SIP server rejects registrations and calls, and
Phone 1000 and Phone 5000 then register with VCX, which accepts their registrations and
calls.
If the link between Router C and VCX is disconnected, Phone 1000 and Phone 5000 can
register with the local SIP server again to originate calls.
184
Configuring the local SIP server to operate in the alive mode
(method 2)
Network requirements
Router A and Router B carry out call services through the VCX. Configure Router A to operate in the
local SIP server alive mode. When the IP link between Router A and VCX fails, the local SIP server
on Router A accepts registrations from its attached phones to make sure the phones can initiate and
receive calls through Router A. When the IP link between Router A and VCX recovers, the local SIP
server on Router A is disabled, and the attached phones re-register with the VCX. See Figure 54.
Figure 54 Network diagram
Configuration procedure
1.
Configure Router A:
# Configure the Ethernet interface.
<RouterA> system-view
[RouterA] interface ethernet 1/1
[RouterA-Ethernet1/1] ip address 1.1.1.2 255.255.255.0
[RouterA-Ethernet1/1] ip address 2.1.1.2 255.255.255.0 sub
[RouterA-Ethernet1/1] quit
# Configure the router to operate in the alive mode.
[RouterA] voice-setup
[RouterA-voice] sip-server
[RouterA-voice-server] server-bind ipv4 2.1.1.2
[RouterA-voice-server] mode alive-server
[RouterA-voice-server] probe remote-server ipv4 3.1.1.1
[RouterA-voice-server] server enable
# Configure number registration information for the phones.
[RouterA-voice-server] register-user 1000
[RouterA-voice-server-user1000] number 1000
[RouterA-voice-server-user1000] quit
[RouterA-voice-server] register-user 5000
[RouterA-voice-server-user5000] number 5000
[RouterA-voice-server-user5000] return
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
185
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 8/0
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] entity 5000 voip
[RouterA-voice-dial-entity5000] address sip proxy
[RouterA-voice-dial-entity5000] match-template 5000
[RouterA-voice-dial-entity5000] quit
[RouterA-voice-dial] quit
# Configure and enable SIP registration.
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 3.1.1.1
[RouterA-voice-sip] registrar ipv4 2.1.1.2 slave
[RouterA-voice-sip] register-enable on
2.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 5000 pots
[RouterB-voice-dial-entity5000] match-template 5000
[RouterB-voice-dial-entity5000] line 2/0
[RouterB-voice-dial-entity5000] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip proxy
[RouterB-voice-dial-entity1000] match-template 1000
# Configure and enable SIP registration.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 3.1.1.1
[RouterB-voice-sip] registrar ipv4 2.1.1.2 slave
[RouterB-voice-sip] register-enable on
3.
Verify the configurations:
{
When the IP link between Router A and VCX fails, the local SIP server on Router A accepts
registrations from its attached phones to make sure that the phones can initiate and receive
calls through Router A. Execute the display voice sip-server register-user command and
you can see that Phone 1000 and Phone 5000 are successfully registered with the local SIP
server Router A.
[RouterA-voice-sip] display voice sip-server register-user all
user
number
status
address
----------------------------------------------------------------------1000
1000
online
1.1.1.2:5060
5000
{
5000
online
2.1.1.1:5060
When the IP link recovers, the local SIP server on Router A is disabled, and phones
re-register with VCX.
186
Configuring the call authority control
Network requirements
As shown in Figure 55, the DNs for Department A in a company are 1000 through 1999, while those
for Department B are 5000 through 5999. The following restrictions need to be implemented:
•
Phones in Department A and Department B cannot originate external calls.
•
Phone 5000 is not allowed to originate calls to phone 1000.
Figure 55 Network diagram
Configuration procedure
1.
Configure Router C:
# Configure the router to operate in the alone mode.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] sip-server
[RouterC-voice-server] server-bind ipv4 2.1.1.2
[RouterC-voice-server] server enable
# Configure authentication information for phones.
[RouterC-voice-server] register-user 1000
[RouterC-voice-server-user1000] number 1000
[RouterC-voice-server-user1000] authentication username 1000 password simple 1000
[RouterC-voice-server-user1000] quit
[RouterC-voice-server] register-user 1111
[RouterC-voice-server-user1111] number 1111
[RouterC-voice-server-user1111] authentication username 1111 password simple 1111
[RouterC-voice-server-user1111] quit
[RouterC-voice-server] register-user 5000
[RouterC-voice-server-user5000] number 5000
[RouterC-voice-server-user5000] authentication username 5000 password simple 5000
[RouterC-voice-server-user5000] quit
[RouterC-voice-server] register-user 5555
[RouterC-voice-server-user5555] number 5555
[RouterC-voice-server-user5555] authentication username 5555 password simple 5555
[RouterC-voice-server-user5555] quit
# Configure a call rule set.
[RouterC-voice-server] call-rule-set
[RouterC-voice-server-set] service 0
187
[RouterC-voice-server-set-svc0] rule 0 deny outgoing any
[RouterC-voice-server-set-svc0] rule 1 permit outgoing 5...
[RouterC-voice-server-set-svc0] rule 2 permit outgoing 1...
[RouterC-voice-server-set-svc0] quit
[RouterC-voice-server-set] service 2
[RouterC-voice-server-set-svc2] rule 0 deny outgoing 1000
[RouterC-voice-server-set-svc2] quit
[RouterC-voice-server-set] quit
[RouterC-voice-server]
# Apply the call rule set.
[RouterC-voice-server] srs 0
[RouterC-voice-server] register-user 5000
[RouterC-voice-server-user5000] srs 2
2.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 2/0
[RouterA-voice-dial-entity1000] user 1000 password simple 1000
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] line 2/1
[RouterA-voice-dial-entity1111] user 1111 password simple 1111
[RouterA-voice-dial-entity1111] quit
[RouterA-voice-dial] entity 5000 voip
[RouterA-voice-dial-entity5000] address sip proxy
[RouterA-voice-dial-entity5000] match-template 5...
[RouterA-voice-dial-entity5000] quit
[RouterA-voice-dial] quit
# Enable SIP registration.
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 2.1.1.2
[RouterA-voice-sip] register-enable on
3.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 5000 pots
[RouterB-voice-dial-entity5000] match-template 5000
[RouterB-voice-dial-entity5000] line 2/0
[RouterB-voice-dial-entity5000] user 5000 password simple 5000
[RouterB-voice-dial-entity5000] quit
[RouterB-voice-dial] entity 5555 pots
[RouterB-voice-dial-entity5555] match-template 5555
188
[RouterB-voice-dial-entity5555] line 2/1
[RouterB-voice-dial-entity5555] user 5555 password simple 5555
[RouterB-voice-dial-entity5555] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip proxy
[RouterB-voice-dial-entity1000] match-template 1...
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] quit
# Enable SIP registration.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 2.1.1.2
[RouterB-voice-sip] register-enable on
4.
Verify the configurations:
[RouterC-voice-server] display voice sip-server register-user all
user
number
status
address
----------------------------------------------------------------------1000
1000
online
1.1.1.1:5060
1111
1111
online
1.1.1.1:5060
5000
5000
online
2.1.1.1:5060
5555
5555
online
2.1.1.1:5060
The four directory numbers are already registered, they cannot originate external calls, and
Phone 5000 cannot originate calls to Phone 1000.
Configuring an area prefix
Network requirements
As shown in Figure 56:
•
The internal numbers of a company are four-digit long and the area prefix is 8899.
•
An external user needs to dial the area prefix 8899 before an internal number. The local SIP
server on Router C removes the area prefix from the dialed number and calls the four-digit
internal number.
•
The external phone attached to Router A is not registered with Router C.
•
The internal phone attached to Router B is registered with Router C.
Figure 56 Network diagram
Configuration procedure
1.
Configure Router C:
# Configure the router to operate in the alone mode.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] sip-server
[RouterC-voice-server] server-bind ipv4 2.1.1.2
[RouterC-voice-server] server enable
189
# Set Router A to a trusted node.
[RouterC-voice-server] trusted-point ipv4 1.1.1.1
# Configure the area prefix 8899.
[RouterC-voice-server] area-prefix 8899
# Configure authentication information for Phone 5000.
[RouterC-voice-server] register-user 5000
[RouterC-voice-server-user5000] number 5000
[RouterC-voice-server-user5000] authentication username 5000 password simple 5000
2.
Configure Router A:
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 55661000 pots
[RouterA-voice-dial-entity55661000] match-template 55661000
[RouterA-voice-dial-entity55661000] line 2/0
[RouterA-voice-dial-entity55661000] quit
[RouterA-voice-dial]entity 88995000 voip
[RouterA-voice-dial-entity88995000] address sip ip 2.1.1.2
[RouterA-voice-dial-entity88995000] match-template 88995000
3.
Configure Router B:
# Configure voice entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 5000 pots
[RouterB-voice-dial-entity5000] match-template 5000
[RouterB-voice-dial-entity5000] user 5000 password simple 5000
[RouterB-voice-dial-entity5000] line 2/0
[RouterB-voice-dial-entity5000] quit
# Enable SIP registration.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 2.1.1.2
[RouterB-voice-sip] register-enable on
4.
Verify the configurations:
{
Execute the display voice sip-server register-user command to see that the directory
number 5000 is registered with the local SIP server Router C.
[RouterC-voice-server-user5000] display voice sip-server register-user all
user
number
status
address
--------------------------------------------------------------------5000
{
5000
online
1.1.1.1:5060
Make a call from the external number 55661000 to the internal number 88995000. The local
SIP server Router C removes the area prefix 8899 from 88995000 to convert it into the
internal short number 5000. Then, the local SIP server alerts Phone 5000, and both parties
can communicate after Phone 5000 is picked up.
190
Configuring a call route
Network requirements
As shown in Figure 57:
•
The internal numbers of a company are four-digit long and the area prefix is 8899.
•
External phone 55665000 attached to Router B is not registered with the local SIP server on
Router C.
•
Internal phone 1000 attached to Router A is already registered with Router C.
When a user in the company dials the external number, the local SIP server will route the call
according to the configured call-out route and add area prefix 8899 to the calling number.
Figure 57 Network diagram
Configuration procedure
1.
Configure Router C:
# Configure the router to operate in the alone mode.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] sip-server
[RouterC-voice-server] server-bind ipv4 2.1.1.2
[RouterC-voice-server] server enable
# Configure a call route.
[RouterC-voice-server] call-route
[RouterC-voice-server-route] trunk 0 called-number 55665000 ipv4 2.1.1.1 area-prefix
8899
# Configure authentication information for Phone 1000.
[RouterC-voice-server] register-user 1000
[RouterC-voice-server-user1000] number 1000
[RouterC-voice-server-user1000] authentication username 1000 password simple 1000
2.
Configure Router A:
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 2/0
[RouterA-voice-dial-entity1000] user 1000 password simple 1000
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial]entity 55665000 voip
[RouterA-voice-dial-entity55665000] address sip proxy
[RouterA-voice-dial-entity55665000] match-template 55665000
[RouterA-voice-dial-entity55665000] quit
[RouterA-voice-dial] quit
191
# Enable SIP registration.
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 2.1.1.2
[RouterA-voice-sip] register-enable on
3.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 55665000 pots
[RouterB-voice-dial-entity55665000] match-template 55665000
[RouterB-voice-dial-entity55665000] line 2/0
[RouterB-voice-dial-entity55665000] quit
4.
Verify the configurations:
[RouterC-voice-server-user1000] display voice sip-server register-user all
user
number
status
address
--------------------------------------------------------------------1000
1000
online
1.1.1.1:5060
The directory number 1000 is registered with the local SIP server Router C. When Phone 1000
dials 55665000, the local SIP server routes the call along the call route to Router B and adds
the area prefix 8899 to the calling number. Phone 55665000 is alerted. Both parties can
communicate after Phone 55665000 is picked up.
192
Configuring SIP trunk
This chapter describes how to configure SIP trunk.
Background
As shown in Figure 58, on a typical telephone network, internal calls of the enterprise are made
through the internal PBX, and external calls are placed over a PSTN trunk.
Figure 58 Typical telephone network
With the development of IP technology, many enterprises deploy SIP-based IP-PBX networks as
shown in Figure 59. Internal calls of the enterprise are made by using the SIP protocol, and external
calls are still placed over a PSTN trunk. The problem is that the enterprises have to maintain both the
SIP network and PSTN trunk, which increases the difficulty of network management.
Figure 59 SIP+PSTN network
As more enterprise IP-PBX networks run SIP and more ITSPs use SIP to provide basic voice
communication structures, enterprises urgently need a technology that can connect the enterprise
IP-PBX network to the ITSP over SIP. This technology is called SIP trunk. A typical network diagram
of SIP trunk is shown in Figure 60.
The SIP trunk function can be embedded into the voice gateway or the firewall deployed at the edge
of an enterprise private network. The device providing the SIP trunk function is called the SIP trunk
device, or the SIP trunk gateway.
193
Figure 60 All IP-based network
Features
SIP trunk has the following features:
•
Only one secure and QoS guaranteed SIP trunk link is required between a SIP trunk device and
the ITSP. The SIP trunk link can carry multiple concurrent calls, and the carrier only
authenticates the link instead of each SIP call carried on this link.
•
The internal calls of the enterprise are placed by the enterprise IP-PBX. The outbound calls of
the enterprise are forwarded by the SIP trunk device to the ITSP, and are finally routed to the
PSTN by the device in the ITSP. Enterprises do not need to maintain the PSTN trunk, which
reduces the costs of hardware and maintenance.
•
By setting destination addresses, the enterprise can connect to multiple ITSPs, making full use
of ITSPs all over the world, and save on call costs.
•
With the SIP trunk device deployed, the entire network can use the SIP protocol to better
support IP communication services, such as voice, conference, and instant messaging.
•
A SIP trunk device differs from a SIP proxy server. The SIP trunk device initiates a new call
request to the ITSP on behalf of the user after receiving a call request from the user, and both
the user and the ITSP communicate only with the SIP trunk device. During the forwarding
process, the SIP trunk device forwards both signaling messages and RTP media messages.
Typical applications
The SIP trunk device is deployed between the enterprise IP-PBX and the ITSP. All internal calls are
placed by the enterprise IP-PBX. All outbound calls are forwarded by the SIP trunk device to the
ITSP through the SIP trunk link. Figure 61 shows a typical network diagram for the SIP trunk
technology.
194
Figure 61 SIP trunk network diagram
Protocols and standards
SIP trunk-related protocols and standards are as follows:
•
RFC 3261
•
RFC 3515
•
SIPconnect Technical Recommendation v1.1
Hardware compatibility with SIP trunk
SIP trunk is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
SIP trunk configuration task list
Task
Remarks
Enabling the SIP trunk function
Required.
Creating a SIP server group
Configuring a SIP
server group
Enabling the real-time switching function
Configuring the keepalive and redundancy
functions
Configuring source address binding
195
Required.
Required when there are multiple
servers in a SIP server group.
Optional.
Task
Configuring a SIP trunk
account
Remarks
Configuring a SIP trunk account for
registration
Configuring registration timers for a SIP
trunk account
Binding a SIP server group to the VoIP
voice entity
Configuring call routes
for outbound calls
Required.
Specifying the destination address
Required.
Use one of the three methods.
Specifying the proxy server used for
outbound calls
Configuring call match rules
Optional.
Configuring call routes for inbound calls
Required.
Enabling codec transparent transfer
Optional.
Enabling media flow-around
Optional.
Enabling delayed offer to early offer conversion
Optional.
Enabling codec transcoding
Optional.
Enabling address hiding
Optional.
Enabling call forwarding
Optional.
Enabling call transfer
Optional.
Enabling midcall signaling pass-through
Optional.
Enabling the SIP trunk function
Before using various SIP trunk functions, first enable the SIP trunk function on the SIP trunk device.
Do not use a device enabled with the SIP trunk function as a SIP UA.
To enable the SIP trunk function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enable the SIP trunk
function.
sip-trunk enable
Disabled by default.
Configuring a SIP server group
This section covers the procedures for creating and configuring a SIP server group.
Creating a SIP server group
Use SIP server groups to manage the registrar and call servers. A SIP server group can be
configured with up to five member servers. An index represents the priority of a member server in a
SIP server group. The smaller the index value, the higher the priority. The currently used SIP server
196
is called the current server. Each server in the SIP server group can be the current server, but there
is only one current server at a time.
To create a SIP server group:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Create a SIP server
group and enter SIP
server group view.
server-group group-number
N/A
4.
Add a member
server to the SIP
server group and
configure the server
information.
By default, a SIP server group has no
member server.
address index-number { ipv4
ip-address | dns dns-name } [ port
port-number ] [ transport { udp | tcp
| tls } ] [ url { sip | sips } ]
You can add at most five member
servers to a SIP server group. An index
represents the priority of a member
server in the SIP server group. The
smaller the index value, the higher the
priority.
Optional.
Not specified by default.
5.
6.
Specify a name for
the SIP server
group.
group-name group-name
Configure a
description for the
SIP server group.
description text
The name of a SIP server group
identifies the SIP server group. The
domain name of the carrier server is
usually used as the name of a SIP
server group. If the name of a SIP
server group is not configured, the host
name specified in the assign
command is used to identify the group,
if any. Otherwise, the IP address or
domain name of the current server in
the SIP server group is used to identify
the group.
Optional.
Not configured by default.
Enabling the real-time switching function
With the real-time switching function enabled, if the SIP trunk device receives no response message
or receives response message 408 or 5XX (excluding 502, 504, 505, and 513) after sending
registration requests to the SIP server, the SIP trunk device tries to connect to the member server
with the second highest priority value in the SIP server group, and so on, until it successfully
connects to a SIP server or has tried all the servers in the group.
With the real-time switching function enabled, if the SIP trunk device receives no response message
or receives response message 403, 408 or 5XX (excluding 502, 504, 505, and 513) after initiating a
call, the SIP trunk device tries to connect to the member server with the second highest priority value
in the SIP server group, and so on, until it successfully connects to a SIP server or has tried all the
servers in the group.
To enable the real-time switching function of a SIP server group:
Step
Command
Remarks
1.
System-view
N/A
Enter system view.
197
Step
Command
Remarks
2.
Enter voice view.
voice-setup
N/A
3.
Create a SIP server
group and enter SIP
server group view.
server-group group-number
N/A
Enable the real-time
switching function in
the SIP server
group.
hot-swap enable
Disabled by default.
4.
NOTE:
The real-time switching time is determined by the voip timer voip-to-pots command. For more
information about the voip timer voip-to-pots command, see Voice Command Reference.
Configuring the keepalive and redundancy functions
Use the keepalive function to detect whether the SIP servers in a SIP server group are reachable.
The SIP trunk device selects the current server according to the detect results and the redundancy
mode. If the keepalive function is disabled, the current server is always the one with the highest
priority in the SIP server group.
A SIP server group might operate in one of the following keepalive modes:
•
OPTIONS mode—The SIP trunk device periodically sends OPTIONS messages to detect the
servers. If the SIP trunk device receives response message 408 or 5XX (excluding 502, 504,
505, and 513) from a SIP server after sending an OPTIONS message, it considers the SIP
server unreachable.
•
REGISTER mode—The REGISTER message can be used to detect the SIP servers. If the SIP
trunk device receives response message 408 or 5XX (excluding 502, 504, 505, and 513) from a
SIP server after sending a REGISTER message, it considers the SIP server unreachable.
The redundancy mode of a SIP server group is configured in SIP client view. A SIP server group
operates in one of the following redundancy modes:
•
Parking mode—The SIP trunk device sends the OPTIONS or REGISTER message to the
current server. When the current server is not available, the SIP trunk device selects the
member server with the second highest priority in the SIP server group as the current server
even if the original current server recovers. Before the parking mode is applied, use the
keepalive command to set OPTIONS or REGISTER as the keep-alive mode.
•
Homing mode—The SIP trunk device sends the OPTIONS messages to both the current
server and the member server with the second highest priority in the SIP server group. When
the current server is not available, the SIP trunk device selects the member server with the
second highest priority as the current server. Once the original current server recovers, or a
server with a higher priority than the current server is available in the SIP server group, the SIP
trunk device selects the original current server or the server with the highest priority as the
current server. Before the homing mode is applied, use the keepalive command to set
OPTIONS as the keepalive mode.
To configure the keepalive and redundancy functions for the SIP server group:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Create a SIP server
group and enter SIP
server-group group-number
N/A
198
Step
Command
Remarks
server group view.
4.
Disabled by default.
Enable the keepalive
function and set the
keepalive mode for
the SIP server
group.
keepalive { options [ interval
seconds ] | register }
If the keepalive function is disabled, the
current server is always the one with
the highest priority in the SIP server
group.
5.
Return to system
view.
quit
N/A
6.
Enter SIP client view.
sip
N/A
7.
Configure the
redundancy mode
for the SIP server
group.
redundancy mode { homing |
parking }
Optional.
Parking mode by default.
Configuring source address binding
Use source address binding to specify a source address for signaling and media flows initiated from
private SIP users to a SIP server group. Perform this task by using one of the following methods:
•
Static IP address binding—Uses a static address as the source IP address.
•
Interface binding—Uses the IP address of a source interface as the source IP address. In a
large-scale network, an interface obtains its IP address from a DHCP or PPPoE server. In this
scenario, you can use this method to configure an interface as the source of SIP signaling and
media flows to avoid manual configuration.
This feature is supported by Layer 3 Ethernet interfaces, Layer 3 Gigabit Ethernet interfaces, and
dialer interfaces.
For more information about DHCP, see Layer 3—IP Services Configuration Guide.
For more information about PPPoE, see Layer 2—WAN Configuration Guide.
Configuration prerequisites
•
Configure a SIP trunk account for registration. For more information, see "Configuring a SIP
trunk account for registration."
•
Configure call routes for outbound calls. For more information, see "Configuring call routes for
outbound calls."
Configuration procedure
To configure source address binding:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Create a SIP server
group and enter SIP
server group view.
server-group group-number
N/A
Specify a source IP
address for sent SIP
signaling or media
flows.
source-bind { media | signal }
{ interface-type interface-number |
ipv4 ip-address }
By default, no source address is bound
for SIP signaling and media flows.
4.
The following table describes how source address binding works in different conditions:
199
Condition
Result
•
Configure a source address binding when
ongoing calls exist.
•
A new source address binding for media does not take
effect for ongoing SIP media sessions but takes effect for
subsequent SIP media sessions.
A new source address binding for signal takes effect
immediately for all SIP signaling sessions.
The bound source interface or the
interface whose IP address is set as the
source address is shut down.
The source IP address binding becomes invalid and will not
work until the interface is up. During the shutdown period, the
gateway automatically gets a source IP address for sent
signaling or media flows.
The bound static IP address is removed or
modified, or the bound interface is
removed.
The source IP address binding is removed.
The bound interface is disconnected.
The source IP address binding is cancelled, and restored when
the interface is connected.
Configure a source address binding when
the physical layer or link layer state of the
corresponding interface is down.
The source address binding does not take effect and the
gateway automatically gets a source IP address for packets.
The DHCP lease duration expires and the
bound interface dynamically obtains a new
IP address from the DHCP server
The new IP address will be used as the source IP address.
The SIP registrar is enabled.
The subsequent registration update messages use the source
IP address newly bound for signaling streams to initiate
registration.
Configuring a SIP trunk account
This section describes how to create, enable, and configure a SIP trunk account.
Configuring a SIP trunk account for registration
SIP trunk accounts are used to manage various information allocated to users by the carrier. You can
configure the authentication username, authentication password, host name, host username, and
the associated SIP server group for a SIP trunk account.
To configure a SIP trunk account for registration:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Create a SIP trunk
account and enter
account view.
sip-trunk account index-number
N/A
Enable the SIP trunk
account.
account enable
Assign the host
username allocated
by the ITSP to the SIP
trunk account.
assign contact-user user-name
Not assigned by default.
Assign the host name
allocated by the ITSP
assign host-name host-name
Optional.
4.
5.
6.
200
Optional.
Enabled by default.
Step
Command
Remarks
to the SIP trunk
account.
7.
8.
9.
Not assigned by default.
Associate the SIP
trunk account with a
SIP server group for
registration.
registrar server-group
group-number [ expires time ]
Configure the
authentication
username and
password for the SIP
trunk account.
user username password { simple
| cipher } password
Optional.
Enable the
registration function
for the SIP trunk
account.
register enable
Disabled by default.
Not associated by default.
Not configured by default.
Configuring registration timers for a SIP trunk account
For more information about the registration timers, see "Configuring registration timers."
Configuring call routes for outbound calls
When a SIP trunk account on the private network needs to call an external user, you need to first bind
the SIP trunk account to a VoIP voice entity, and then configure call routes using one of the following
three methods:
•
Binding a SIP server group to the VoIP voice entity
•
Specifying the destination address
•
Specifying the proxy server used for outbound calls
Binding a SIP server group to the VoIP voice entity
After binding a SIP trunk account to a VoIP voice entity, bind a SIP server group to the same VoIP
voice entity to make the SIP trunk account use the SIP server group to place calls.
To bind a SIP server group to the VoIP voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity
view.
entity entity-number voip
N/A
5.
6.
Bind a SIP trunk
account to the VoIP
voice entity.
bind sip-trunk account account-index
Bind a SIP server group
address sip server-group group-number
201
By default, no SIP trunk
account is bound to the VoIP
voice entity.
Only an existing SIP trunk
account can be bound to a
VoIP voice entity.
By default, no SIP server
Step
Command
Remarks
to the VoIP voice entity.
group is bound to the VoIP
voice entity.
A VoIP voice entity can have
only one existing SIP server
group bound to it.
Specifying the destination address
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity
view.
entity entity-number voip
N/A
5.
6.
Bind a SIP trunk account
to the VoIP voice entity.
bind sip-trunk account account-index
Specifying the
destination address.
address sip { dns domain-name [ port
port-number ] | ip ip-address [ port
port-number ] }
By default, no SIP trunk
account is bound to the VoIP
voice entity.
Only an existing SIP trunk
account can be bound to a
VoIP voice entity.
By default, no destination
address is specified for the
VoIP voice entity.
Specifying the proxy server used for outbound calls
When a SIP proxy server is used to route SIP messages and audio traffic, the SIP trunk device uses
the SIP server group specified by the proxy command to make calls.
If no registrar is specified by the registrar or registrar server-group command in SIP client view,
the proxy server or SIP server group specified by the proxy command can be used as the registrar.
To specify the proxy server used for outbound calls:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial
program view.
dial-program
N/A
4.
Enter VoIP voice
entity view.
entity entity-number voip
N/A
5.
Bind a SIP trunk
account to the VoIP
voice entity.
bind sip-trunk account
account-index
Enable the SIP proxy
server.
address sip proxy
6.
202
By default, no SIP trunk account is
bound to the VoIP voice entity.
Only an existing SIP trunk account can
be bound to a VoIP voice entity.
Not configured by default.
Step
Command
Remarks
7.
Return to system
view.
quit
N/A
8.
Enter SIP client view.
sip
N/A
9.
Specify a SIP server
group to be used as
the proxy server.
proxy { dns domain-name | ipv4
ip-address [ port port-number ] |
server-group group-number }
Not configured by default.
Configuring call match rules
You can control call route selection by configuring the prefix of the source host name, prefix of the
destination host name, or the source IP address as the call match rule in VoIP voice entity view. If you
have configured several call match rules, only the callings that match all rules are permitted.
Configure the call match rules as follows:
•
Use the match source host-prefix command to specify the prefix of source host name as a
call match rule for a VoIP voice entity. The specified prefix of source host name is used to match
against the source host names of calls.
{
{
{
If the INVITE message received by the SIP trunk device carries the Remote-Party-ID
header, the calling information is abstracted from this header field.
If the INVITE message received by the SIP trunk device carries the Privacy header, the
source host name is abstracted from the P-Asserted-Identity or P-Preferred-Identity header
field.
If the INVITE message received by the SIP trunk device does not carry any of the previously
mentioned three header fields, the host name in the From header field of the INVITE
message is used as the source host name.
•
Use the match destination host-prefix command to specify the prefix of destination host
name as a call match rule for a VoIP voice entity. The specified prefix of destination host name
is used to match against the destination host names of calls. The host name in the To header
field of an INVITE message received by the SIP trunk device is used as the destination host
name.
•
Use the match source address command to specify the source address as a call match rule
for a VoIP voice entity.
To configure the call match rules:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter VoIP voice entity
view.
entity entity-number voip
N/A
5.
Match a source host
name prefix for the VoIP
voice entity.
match source host-prefix host-prefix
Match a destination host
name prefix for the VoIP
voice entity.
match destination host-prefix
host-prefix
6.
Optional.
Not specified by default. In
other words, all source host
names can be matched.
Optional.
203
Not specified by default. In
other words, all destination
host names can be matched.
Step
Command
7.
match source address { ipv4 ip-address
| dns dns-name | server-group
group-index }
Match a source address
for the VoIP voice entity.
Remarks
Optional.
Not specified by default. In
other words, all source
addresses can be matched.
Configuring call routes for inbound calls
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
5.
Configure the called number
template for the VoIP voice
entity.
match-template match-string
Not configured by default.
Configure the call route.
address sip { dns domain-name
[ port port-number ] | ip ip-address
[ port port-number ] | proxy }
Not configured by default.
6.
Enabling codec transparent transfer
If the SIP trunk device does not support the codec capability sets supported by the calling and called
parties, enable codec transparent transfer on the SIP trunk device. The SIP trunk device
transparently transfers codec capability sets between two parties. The calling and called parties
complete the codec negotiation.
To enable codec transparent transfer:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
Optional.
5.
Enable codec transparent
transfer.
codec transparent
By default, codec
transparent transfer is
disabled and the SIP trunk
device is involved in the
media negotiation
between the calling and
called parties.
Note that to enable codec
transparent transfer on the
SIP trunk device, execute
this command on all VoIP
voice entities connected to
the public and private
204
Step
Command
Remarks
networks.
Enabling media flow-around
This function enables the media packets to pass directly between two SIP endpoints, without the
intervention of the SIP trunk device. The media packets flow around the SIP trunk device.
To enable media flow-around:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
Optional.
5.
Enable media flow-around.
media flow-around
By default, media
flow-around is disabled.
The SIP trunk device acts
as the RTP trunk proxy
and forwards the media
packets.
Enabling delayed offer to early offer conversion
An INVITE message sent with SDP in the message body defines an early offer, and an INVITE
message sent without SDP in the message body defines a delayed offer. Some carriers mandate
early offer calls for charge security. To meet this requirement, enable delayed offer to early offer
(DO-EO) conversion on the SIP trunk device.
To enable DO-EO conversion:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
5.
Enable DO-EO conversion.
early-offer forced
Optional.
By default, DO-EO
conversion is disabled.
NOTE:
If codec transparent transfer or media flow-around is enabled, the early-offer forced command
does not takes effect.
205
Enabling codec transcoding
In the scenario where the SIP trunk device controls the results of media capability negotiation, if the
SIP trunk device cannot find a common codec for two parties during negotiation, the two parties will
fail to establish a call. In this case, enable codec transcoding on the SIP trunk device.
With this function enabled, the SIP trunk device uses its own codec capability set to negotiate with
the calling and called parties respectively. If the negotiated codecs with the two parties do not match,
the SIP trunk device transcodes the media flows passing through it.
The codec transcoding feature does not take effect in any of the following cases:
•
Codec transcoding is enabled, but no DSP resources are available for codec transcoding.
•
Codec transparent transfer is enabled.
•
Media flow-around is enabled.
To enable codec transcoding:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
5.
Enable codec transcoding.
codec transcoding
Optional.
By default, codec
transcoding is disabled.
Enabling address hiding
If the SIP trunk device acts as the session border gateway, the addresses of the endpoints are
always hidden to the peers. The SIP trunk device replaces the endpoints' addresses carried in SIP
messages with the addresses of the corresponding egress interfaces. For both the calling and called
parties, the SIP trunk device is the only ingress of signaling and media packets.
Address hiding does not take effect for a voice entity enabled with media flow-around.
To enable address hiding:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
Optional.
4.
Enable address hiding.
address-hiding enable
206
By default, address hiding
for SIP-to-SIP calls is
disabled.
Enabling call forwarding
Usually, the SIP trunk device transparently transfers the SIP messages carrying call forwarding
information to the endpoints, and the endpoints perform the call forwarding. However, some
endpoints do not support call forwarding information. In this case, enable the call forwarding function
on the SIP trunk device.
With this function enabled, the SIP trunk device processes call forwarding information without
notifying the calling parties, and does not transfer call forwarding information to the endpoints.
To enable call forwarding:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
5.
Enable call forwarding.
supplementary−service sip
call-forwarding
Optional.
By default, this feature is
disabled.
Enabling call transfer
Usually, the SIP trunk device transparently transfers the SIP messages carrying call transfer
information to the endpoints, and the endpoints perform the call transfer. However, some endpoints
do not support call transfer information. In this case, enable the call transfer function on the SIP trunk
device.
With this function enabled, the SIP trunk device processes call transfer information without notifying
the calling parties, and does not send call transfer information to the endpoints.
To enable call transfer:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
5.
Enable call transfer.
supplementary−service sip
call-transfer
Optional.
By default, call forwarding
is disabled.
Enabling midcall signaling pass-through
When the SIP trunk device forwards the media flows between the calling and called parties, the
device can monitor the status of the media flows over a period of time and determine whether the
endpoints have hung up abnormally. If media flow-around is enabled, the SIP trunk device does not
process the media packets from endpoints, and cannot monitor the status of the calling and called
parties.
207
To solve this problem, the involved endpoints can use the session timer mechanism to send periodic
re-INVITE or UPDATE requests to notify the SIP trunk device about session state.
By default, the session timer mechanism is initiated by the calling party. If the called party also
supports this mechanism, you can enable mid-call signaling pass-through to enable the called party
to process the session update information. Otherwise, the session timer mechanism only works
between the calling party and the SIP trunk device. The interval for sending session update requests
is negotiated by endpoints.
To enable midcall signaling pass-through:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Enter VoIP voice entity view.
entity entity-number voip
N/A
Optional.
5.
Enable mid-call signaling
pass-through.
By default, the mid-call
signaling pass-through of
SIP-to-SIP calls is
disabled.
midcall-signal passthrough
Displaying and maintaining SIP trunk
configuration
Task
Command
Display SIP trunk account
information.
display voice sip-trunk account
[ | { begin | exclude | include }
regular-expression ]
Display SIP server group
information.
display voice server-group
[ group-number ] [ | { begin |
exclude | include }
regular-expression ]
Remarks
Available in any view.
SIP trunk configuration examples
This section provides configuration examples of SIP trunk.
Configuring a SIP server group with only one member server
Network requirements
As shown in Figure 62, the enterprise private network has a SIP trunk device deployed. Router A is a
private network device, and Router B is a public network device. All calls between the private
network and public network are made through the SIP trunk device.
208
Figure 62 Network diagram
Configuration procedure
1.
Configure Router A:
# Specify 2000 as a local number of POTS voice entity 2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 pots
[RouterA-voice-dial-entity2000] line 1/0
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
# Configure VoIP voice entity 1000: the called number is 1000, and the IP address of the peer
end is 1.1.1.2 (the address of the interface on the SIP trunk device).
[RouterA-voice-dial] entity 1000 voip
[RouterA-voice-dial-entity1000] address sip ip 1.1.1.2
[RouterA-voice-dial-entity1000] match-template 1000
2.
Configure the SIP trunk device:
# Enable the SIP trunk function.
<TG> system-view
[TG] voice-setup
[TG-voice] sip-trunk enable
# Create SIP server group 1. Add a SIP server into the server group: the index and the IPv4
address of the server are 1 and 10.1.1.2 respectively.
[TG-voice] server-group 1
[TG-voice-group1] address 1 ipv4 10.1.1.2
[TG-voice-group1] quit
# Create SIP trunk account 1 with the host user name 2000, and associate the account with SIP
server group 1.
[TG-voice] sip-trunk account 1
[TG-voice-account-1] assign contact-user 2000
[TG-voice-account-1] registrar server-group 1
[TG-voice-account-1] register enable
[TG-voice-account-1] quit
209
# Configure the call route for the outbound calls from private network user 2000 to public
network user 1000 by binding SIP server group 1 to VoIP voice entity 1.
[TG-voice] dial-program
[TG-voice-dial] entity 1 voip
[TG-voice-dial-entity1] address sip server-group 1
[TG-voice-dial-entity1] bind sip-trunk account 1
[TG-voice-dial-entity1] match-template 1000
[TG-voice-dial-entity1] quit
# Configure the call route for the inbound calls from public network user 1000 to private network
user 2000. Configure the IP address of the peer end as 1.1.1.1, which is the address of the
interface on Router A.
[TG-voice-dial] entity 2 voip
[TG-voice-dial-entity2] address sip ip 1.1.1.1
[TG-voice-dial-entity2] match-template 2000
3.
Configure Router B:
# Specify 1000 as a local number of POTS voice entity 2000.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 pots
[RouterB-voice-dial-entity1000] line 1/0
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
# Configure VoIP voice entity 2000. The called number is 2000. Specify to use the SIP proxy
server for SIP message exchange.
[RouterB-voice-dial] entity 2000 voip
[RouterB-voice-dial-entity2000] address sip proxy
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] quit
[RouterB-voice-dial] quit
# Configure the IPv4 address of the registrar as 10.1.1.2 and enable the registrar.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 10.1.1.2
[RouterB-voice-sip] register-enable on
Cofngiuration verification
1.
On the SIP trunk device, display SIP trunk account information.
[TG-voice-dial-entity2] display voice sip-trunk account
ID User
Group Server
Exp
Status
1
1
1802
Online
2000
10.1.1.2:5060
The output shows that the private network account 2000 has registered with the server at
10.1.1.2.
2.
All calls between the private network and public network are made through the SIP trunk
device.
Execute the display voice statistics call-active command on the SIP trunk device and you
can see that all calls between the private network and public network are made through the SIP
trunk device.
3.
On the SIP server of the carrier, you can view only the interface address of the SIP trunk device,
which means that the SIP trunk device can filter the information of the enterprise private
network users.
210
Configuring a SIP server group with multiple member servers
Network requirements
As shown in Figure 63, the enterprise private network has a SIP trunk device. Router A is a private
network device, and Router B is a public network device. All calls between the private network and
public network are made through the SIP trunk device. The carrier is required to provide multiple
servers to ensure call reliability.
Figure 63 Network diagram
Configuration procedure
# Enable the SIP trunk function.
<TG> system-view
[TG] voice-setup
[TG-voice] sip-trunk enable
# Create SIP server group 1. Add two SIP servers into the server group: the IP addresses are
10.1.1.2 and 10.1.1.3, and the server with the address 10.1.1.2 has a higher priority value.
[TG-voice] server-group 1
[TG-voice-group-1] address 1 ipv4 10.1.1.2
[TG-voice-group-1] address 3 ipv4 10.1.1.3
# Enable the real-time switching function of SIP server group 1. Set the keepalive mode and
redundancy mode for SIP server group 1 to OPTIONS and parking respectively.
[TG-voice-group-1] keepalive options
[TG-voice-group-1] hot-swap enable
[TG-voice-group-1] quit
[TG-voice] sip
[TG-voice-sip] redundancy mode parking
Other configurations on the SIP trunk device and other devices are the same as those described in
"Configuration procedure."
211
Configuration verification
1.
When the SIP server with IP address 10.1.1.2 fails, the SIP server with IP address 10.1.1.3
takes over communications between the private network and the public network. After that, the
communications recover.
2.
After the SIP server with IP address 10.1.1.2 recovers, it does not take over call processing and
the current SIP server with IP address 10.1.1.3 keeps working.
Configuring call match rules
Network requirements
As shown in Figure 64, the enterprise private network has a SIP trunk device deployed. Router A1
and Router A2 are private network devices, and Router B is a public network device. Users
connected to Router A2 are not allowed to call public network users. All calls between the private
network and public network are made through the SIP trunk device.
Figure 64 Network diagram
Configuration procedure
# Configure the call route for the outbound calls from private network user 2000 to public network
user 1000 by binding SIP server group 1 to VoIP voice entity 1.
<TG> system-view
[TG] voice-setup
[TG-voice] sip-trunk enable
[TG-voice] dial-program
[TG-voice-dial] entity 1 voip
[TG-voice-dial-entity1] address sip server-group 1
[TG-voice-dial-entity1] bind sip-trunk account 1
[TG-voice-dial-entity1] match-template 1000
# Specify that calls with the source IP addresses 1.1.1.1 are permitted on VoIP voice entity 1.
[TG-voice-dial-entity1] match source address ipv4 1.1.1.1
Other configurations on the SIP trunk device and on other devices are the same as that described in
"Configuration procedure."
Configuration verification
•
Private network users connected to Router A1 can call public network users, but private
network users connected to Router A2 cannot call public network users.
212
•
Public network users can call any private network user.
213
Configuring H.323
H.323 is an application-layer control protocol for establishing and terminating multimedia sessions
with one or more participants. H.323 can dynamically adjust and modify session attributes, such as
required session bandwidth, media type (voice and video), encoding/decoding format of media, and
support for broadcast.
H.323 uses the Client/Server model to establish calls through communication between the gateway
and gatekeeper.
H.323, a packet-based multimedia communications system, is an ITU-T standard that specifies the
components, protocols, and procedures that provide multimedia communication services over an IP
network that does not provide guaranteed quality of service (QoS). As one of the standards for VoIP,
H.323 has long been used by traditional carriers and network device manufacturers in their VoIP
solutions.
As shown in Figure 65, the H.323 stack is implemented at the application layer. The H.323 standard
includes these protocols:
•
H.225.0, including Q.931 and registration, administration and status (RAS) protocol, and H.245
for signaling control
•
G.711, G.729, G.723.1, and G.723.A for audio codec
•
H.261 and H.263 for video codec
•
T.120 series (including T.123, T.124, T.125, T.126, T.127, and T.324) for multimedia data
transmission.
The real-time transport protocol (RTP) provides end-to-end real-time audio and video delivery
services. Its functionality is enhanced through its control protocol, the real-time transport control
protocol (RTCP). The primary function of RTCP is to provide feedback on the quality of data
distribution, which allows the application system to accommodate to different network conditions and
helps with fault location and diagnosis as well. These two protocols work together to ensure real-time
voice transmission.
Figure 65 illustrates the H.323 stack:
Figure 65 H.323 stack
Data
T.126
Signaling
T.127
T.126 T.324 T.127
T.124
T.324
T.125
T.124
T.123
T.125
H.245
H.225.0
H.245
RAS
H.225.0
RAS
T.123TCP
TCP
Audio
Video
G.711
G.729
H.261
G.711
G.723.1
H.263
G.729
H.261
G.723.A
G.723.1
H.263
G.723.A
RTP RTCP
UDP
RTP RTCP
Network layer
UDP
Link
layer
网络层
Physical
layer
链路层
ITU-T RAS is implemented in compliance with the H.323v2 protocol for communication between
gateway and gatekeeper. RAS adopts the communication model where a gateway sends requests
and the gatekeeper responds with confirms or rejects listed in the following table.
214
Table 16 Major RAS messages
Category
Message
RRQ (Registration_Request)
Registration
RCF (Registration_Confirm)
RRJ (Registration_Reject)
URQ (Unregister_Request)
Unregistration
UCF (Unregister_Confirm)
URJ (Unregister_Reject)
MRQ (Manage_Request)
Management
MCF (Manage_Confirm)
MRJ (Manage_Reject)
ARQ (Admission_Request)
Admission
ACF (Admission_Confirm)
ARJ (Admission_Reject)
LRQ (Location_Request)
Location requests and responses
LCF (Location_Confirm)
LRJ (Location_Reject)
DRQ (Disengage_Request)
Call disengage
DCF (Disengage_Confirm)
DRJ (Disengage_Reject)
IRQ (Information_Request)
Status information
IRR (Information_Request_Response)
IACK (Info_Request_Acknowledge)
INAK (Info_Request_Neg_Acknowledge)
BRQ (Bandwidth_Request)
Bandwidth control
BCF (Bandwidth_Confirm)
BRJ (Bandwidth_Reject)
Gateway resource availability
RAS timer modification
RAI (Resource Availability Indicator)
RAC (Resource Availability Confirm)
RIP
H.323 architecture
An H.323 network consists of terminals, voice gateways, an optional gatekeeper, and multipoint
control units (MCUs). If a gatekeeper is present in an H.323 network, the terminals, gateways, and
MCUs registered with the gatekeeper form a zone. In this zone, the gatekeeper provides the
following functions for H.323 terminals, gateways, or MCUs:
•
Address translation
•
Admission control
•
Bandwidth control
•
Zone administration and security control
•
Call control signaling and call management
•
Routing control and accounting
215
Figure 66 shows a simple H.323 network. For all calls, a gatekeeper is the call service control and
central control unit in its administrative zone.
Gateway entities are usually deployed on routers. Configure the IP voice gateway function on the
router at CLI. They interact with the gatekeeper by sending H.225.0 RAS messages.
In the current implementation, the gatekeeper is usually deployed on SUN stations or servers while
gateways are deployed on routers. For the reliability’s sake, a backup gateway is configured. In the
event communication with a primary gatekeeper becomes abnormal (for example, timeout occurs) or
the primary gatekeeper becomes unavailable, gateways can turn to the backup gatekeeper for
registration.
Figure 66 H.323 network
IP network
Telephone
Gateway
Gateway
Telephone
Gatekeeper
H.323 fundamentals
This section provides a brief description of how an H.323 network functions.
Gatekeeper discovery
When an endpoint wants to establish a call with another endpoint, it first looks for a gatekeeper with
which it can register for services. This process is called gatekeeper discovery. An RAS signaling
message exchange occurs between endpoints and the gatekeeper. The calling endpoint unicasts or
broadcasts a Gatekeeper_Request (GRQ) message. After receiving the request, the gatekeeper
replies with a Gatekeeper_Confirm (GCF) or Gatekeeper_Reject (GRJ) message.
Registration
When the calling endpoint receives a GCF message from the gatekeeper, it sends a
Registration_Request (RRQ) message, requesting to join the zone governed by the gatekeeper. The
gatekeeper replies with a Registration_Confirm (RCF) or Registration_Reject (RRJ) message to
accept or reject the registration request.
After registration, either the endpoint or the gatekeeper sends an Unregister_Request (URQ)
message. However, it is up to the gatekeeper to determine whether to cancel registration, while the
endpoint can only reply with an Unregister_Confirm (UCF) message to cancel registration.
Address translation
If the calling endpoint knows the alias, but not the call signaling address of the called endpoint, it
sends a Location_Request (LRQ) message to the gatekeeper.
Admission control
If the address of the called endpoint is available, the calling endpoint sends an Admission_Request
(ARQ) message, based on which the gatekeeper decides whether to allow this endpoint to join a call
216
process. This is how the gatekeeper controls admission. In the ARQ message sent to the gatekeeper,
the calling endpoint might ask for direct call signaling (see Figure 67) or gatekeeper-routed call
signaling (see Figure 68). The gatekeeper finalizes the way of sending call signaling to the called
endpoint and notifies the calling endpoint of it through an Admission_Confirm (ACF) message.
Figure 67 Direct call signaling between endpoints
Endpoint 1
GateKeeper 1
Endpoint 2
(1) ARQ
(2) ACF/ARJ
(3) Setup
(4) Call Proceeding
(5) ARQ
(6) ACF/ARJ
(7) Alerting
(8) Connect
RAS Messages
Call Signaling Messages
Figure 68 Gatekeeper-routed call signaling
217
Call setup
After receiving the ACF message from the gatekeeper, the calling endpoint sends call signaling to
set up a call. In a direct call signaling for example, the calling endpoint first sends a Setup message
to the called endpoint requesting for a connection.
Call proceeding
After receiving the Setup message, the called endpoint replies with a Call Proceeding message.
However, it might not send this message.
Alerting
Then the called endpoint might send an Alerting message to the calling endpoint, indicating its status
(for example, ringing). The endpoint might not send this message.
Connection
If the called endpoint accepts the call, it must send a Connect message.
Capability negotiation
After the calling endpoint receives the connect message, an H.245 control signaling channel is
established to control the media session between the endpoints. First, the two endpoints exchange
information on their capabilities, such as media format.
Opening/closing logical channels
The two endpoints open one or more logical channels between them for transporting media streams.
(The logical channels are specified by IP address plus port number.)
These channels are closed when the communication is over.
Complete release
Finally, either endpoint in communication can release resources by sending a Release Complete
message.
Disconnection
The endpoints each send a Disconnect_Request (DRQ) to their own gatekeepers, which will confirm
or reject the request. The gatekeepers, however, might send a DRQ to the endpoints. In this case,
the endpoints can only confirm the request.
218
Figure 69 Call setup flow and disconnection flow in which gatekeepers are involved
Hardware compatibility with H.323
H.323 is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
H.323 gateway configuration task list
Task
Remarks
Configuring basic H.323 gateway functions
Required.
Configuring registration password
Optional.
Enabling security calling
Optional.
219
Configuring basic H.323 gateway functions
Configuration prerequisites
Complete the required configuration of POTS and VoIP voice entities.
Configuration guidelines
•
The gatekeeper identifies the type of a gateway by its area ID. The gatekeeper and gateways
reach an agreement on related gateway types in advance. For example, Area ID 1# represents
a voice gateway, and Area ID 2# represents a video gateway. Before a VoIP entity
communicates with the gatekeeper, the gatekeeper judges the gateway type by the received
area ID. At most 30 area IDs can be configured in GK-Client view.
•
Before you can configure a secondary gatekeeper, configure a primary gatekeeper with the
gk-id command.
•
Use the ras-on command to register with the gatekeeper only after you complete all required
basic voice gateway configurations.
Configuration procedure
To configure basic H.323 gateway functions:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
Optional.
The default is Wqldg0Hcwfydz.
3.
Configure an H.323
descriptor.
voip h323-descriptor descriptor
If you want to change the
descriptor, make sure you
understand the impact of this
command before you use it.
4.
Enter gateway view.
gk-client
N/A
5.
Assign an area ID to the
H.323 gateway.
area-id string
Bind a source IP address
(typically the IP address of a
loopback interface) with the
H.323 gateway.
gw-address ip-address
No source IP address is bound by
default.
7.
Configure a gateway alias.
gw-id namestring
No gateway alias is configured by
default. Each gateway can be
assigned only one alias.
8.
Configure a primary
gatekeeper.
gk-id gk-name gk-addr
gk-ipaddress [ ras-port ]
No primary gatekeeper is
configured by default.
9.
Configure a secondary
gatekeeper.
gk-2nd-id gk-name gk-addr
gk-ipaddress [ ras-port ]
6.
10. Enable the gatekeeper client
function to register with the
gatekeeper.
ras-on
Optional.
No area ID is assigned by default.
Optional.
No secondary gatekeeper is
configured by default.
Disabled by default.
For more information about POTS and VoIP entities, see "Configuring voice entities."
220
Configuring registration password
You can configure the gateway to send a password with the RPQ message sent for registration with
the gatekeeper. When the gatekeeper receives this request, it compares the password with the one
configured on it for the gateway. The gatekeeper will accept the request and return an RCF only
when they are the same.
After the GK Client (router) is configured with the registration password, the password is carried
during the whole registration process.
Configuration prerequisites
Complete the required basic H.323 gateway configurations except for the ras-on command. (This
command is used for initiating registration requests to the gatekeeper after all configurations are
completed.)
Configuration procedure
To configure a registration password:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter gateway view.
gk-client
N/A
4.
Configure a gateway
registration password.
gk-security register-pwd
{ cipher | simple } password
No registration password is
configured by default.
Enabling security calling
To let the gateway pass call tokens from the gatekeeper, configure security calling.
After a call is originated, the originating gateway obtains a call token from the originating gatekeeper
and transparently transmits it to the terminating gateway which will then pass the token to the
terminating gatekeeper. Tokens can be synchronized among gatekeepers. The terminating
gatekeeper can return the call accept message to the terminating gateway only after it accepts the
token.
Configuration prerequisites
Complete the required basic H.323 gateway configurations except for the ras-on command. (This
command is used for initiating registration requests to the gatekeeper after all configurations are
completed.)
Configuration procedure
To enable security calling:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter gateway view.
gk-client
N/A
4.
Enable security calling.
gk-security call enable
Enabled by default.
221
NOTE:
Security calling must be disabled in a voice network where the terminating gatekeeper has no ability
to process a call token.
Displaying and maintaining the H.323 gateway
Task
Command
Remarks
Display the registration state
information of a gateway.
display voice gateway [ | { begin
| exclude | include }
regular-expression ]
Available in any view.
H.323 gateway configuration example
Network requirements
Telephones in City A and City B can communicate with each other through routers with the voice
function across an IP network where a gatekeeper is used for dynamic telephone number to IP
address translation.
On City A Router:
•
A loopback interface is used as an H.323 gateway interface and assigned the IP address of
1.1.1.1.
•
The gateway alias is citya-gw.
•
The name of the gatekeeper is gk-center and the address is 3.3.3.3.
•
The RAS port is 1719 and the area ID is 1#.
On City B router:
•
A loopback interface is used as an H.323 gateway interface and assigned the IP address of
2.2.2.2.
•
The gateway alias is cityb-gw.
•
Other configurations are the same as those at City A end.
Figure 70 Network diagram
Configuration procedure
1.
Configure Router A:
# Create a VoIP entity.
<RouterA> system-view
[RouterA] voice-setup
222
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address ras
[RouterA-voice-dial-entity755] quit
# Create a POTS entity.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/1
# Specify a loopback interface as the H.323 gateway interface.
[RouterA-voice-dial-entity1001] return
<RouterA> system-view
[RouterA] interface loopback 0
[RouterA-Loopback0] ip address 1.1.1.1 255.255.255.255
# Enter gateway client view.
[RouterA-Loopback0] quit
[RouterA] voice-setup
[RouterA-voice] gk-client
# Configure the gateway alias, and the name and IP address of the gatekeeper.
[RouterA-voice-gk] gw-address 1.1.1.1
[RouterA-voice-gk] gw-id citya-gw
[RouterA-voice-gk] gk-id gk-center gk-addr 3.3.3.3 1719
# Configure the area ID.
[RouterA-voice-gk] area-id 1#
# Originate registration to the GK.
[RouterA-voice-gk] ras-on
2.
Configure Router B:
# Create a VoIP entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address ras
[RouterB-voice-dial-entity10] quit
# Create a POTS entity.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/1
# Specify a loopback interface as the H.323 gateway interface.
[RouterB-voice-dial-entity2001] return
<RouterB> system-view
[RouterB] interface loopback 0
[RouterB-Loopback0] ip address 2.2.2.2 255.255.255.255
# Enter gatekeeper client view.
[RouterB-Loopback0] quit
[RouterB] voice-setup
223
[RouterB-voice] gk-client
# Configure the gateway alias, and the name and IP address of the gatekeeper.
[RouterB-voice-gk] gw-address 2.2.2.2
[RouterB-voice-gk] gw-id cityb-gw
[RouterB-voice-gk] gk-id gk-center gk-addr 3.3.3.3 1719
# Configure the area ID.
[RouterB-voice-gk] area-id 1#
# Originate registration to the GK.
[RouterB-voice-gk] ras-on
Troubleshooting H.323 gateway
Registration failure
Symptom
The gateway failed to register with the gatekeeper.
Solution
Check that:
1.
The gateway and the gatekeeper can communicate with each other at the network layer with
the ping command.
2.
Execute the display current-configuration command to make the ras-on command takes
effect.
3.
The gatekeeper function is activated at the gatekeeper end.
4.
The area ID is configured on the gatekeeper.
5.
If the node has been logged off, execute the ras-on command on the gateway to re-register
with the gatekeeper.
224
Configuring call services
More and more VoIP-based services are demanded as voice application environments expand. On
the basis of basic calls, new features are implemented to meet different application requirements of
VoIP subscribers.
Call waiting
When subscriber C calls subscriber A who is already engaged in a call with subscriber B, the call will
not be rejected if call waiting is enabled. Just like a normal call, subscriber C will hear ringback tones,
while subscriber A will hear call waiting tones as a reminder that a call is waiting on the line.
Subscriber A can answer the new call by pressing the flash hook or hanging up to end the call with
subscriber B. In the former case, subscriber B is held. In the latter case, subscriber A is immediately
alerted and can pick up the phone to answer the call originated by subscriber C (the waiting call).
Call hold
If subscriber A in a conversation with subscriber B presses the flash hook, the media session of
subscriber B is temporarily cut through and is held (in silent state or listening to waiting tones). The
system plays silent tones or dial tones to subscriber A, depending on the configuration. (The system
first plays dial tones and waits for the subscriber to dial. If the subscriber fails to dial within a period of
time, the system stops playing dial tones and the line stays on hold.). Subscriber A can resume the
call with subscriber B by pressing the flash hook again.
After pressing the flash hook, subscriber A hears dial tones and can initiate a new call. The setup
flow for the new call is the same as the one for ordinary calls.
Call forwarding
When the called party cannot answer the call, after receiving a session request, the called party
notifies in a response the calling party of the forwarded-to number. Then the calling party re-initiates
a session request to the new destination. This is call forwarding.
The system supports the following types of call forwarding:
•
Call forwarding unconditional—With this feature enabled on a voice subscriber line,
incoming calls will be forwarded to the predetermined destination, whether or not the voice
subscriber line is available.
•
Call forwarding busy—With this feature enabled on a voice subscriber line, an incoming call
will be forwarded to the predetermined destination when the voice subscriber line is busy.
•
Call forwarding no reply—With this feature enabled on a voice subscriber line, an incoming
call will be forwarded to the predetermined destination when the voice subscriber line is not
answered within a period of time. The time period is configured by executing the timer
ring-back command and defaults to 60 seconds.
•
Call forwarding unavailable—With this feature enabled on a voice subscriber line, an
incoming call will be forwarded to the predetermined destination when the voice subscriber line
is shut down by executing the shutdown command.
Call transfer
Subscriber A (originator) and subscriber B (recipient) are in a conversation. Subscriber A presses the
flash hook and the call is put on hold. Subscriber A dials another number to originate a call to
225
subscriber C (final recipient). After Subscriber A hangs up, the call between subscriber B and
subscriber C is established. This is call transfer.
To perfect the call transfer feature, the device supports the call recovery function after the call
transfer fails. In other words, if subscriber C in the previous example is in a conversation with another
subscriber and cannot establish a conversation with subscriber B, the call between subscriber A and
subscriber B is recovered.
Call backup
This call service is used when, after initiating a call to the called party, the calling party is unable to
receive a response, or the VoIP device of the called party denied the call request from the calling
party. In this case, if there is another link (PSTN link or VoIP link) to the called party, the calling party
re-initiates a call to the called party over the new route. This is call backup.
Hunt group
Multiple voice subscriber lines are configured with the same called number to form a hunt group. If
the voice subscriber line with the first priority is unavailable when a call setup request to the called
party is received, the call will still be established through another voice subscriber line in the hunt
group.
Call barring
Call barring includes incoming call barring and outgoing call barring.
Incoming call barring usually refers to the Do Not Disturb (DND) service. When incoming call barring
is enabled on a voice subscribe line, calls originated to the attached phone will fail.
When outgoing call barring is enabled on a voice subscriber line, calls originated from the attached
phone will fail, too.
Message waiting indication
The message waiting indication (MWI) feature allows a voice gateway to notify a subscriber of
messages received from a voice mailbox server. For example, when a call destined to subscriber A is
forwarded to the voice mailbox server, the server will notify the state change to the voice gateway.
When subscriber A picks up the phone, subscriber A will hear the message waiting tone without
needing to query the mailbox.
Three-party conference
When subscriber A has a call with subscriber B and holds a call with subscriber C, A can make C join
the current conversation to implement a three-party conference.
During a three-party conference, a passive participant can initiate a new call to create another
conversation. In this way, conference chaining is implemented, and each conference initiator serves
as a conference bridge.
Silent monitor and barge in services
Silent monitor service—Allows a supervisor to monitor active calls without being heard.
Barge in service—Allows a supervisor to participate in a monitored call, thus implementing
three-party conference. For example, subscribers A and B are in a conversation, and subscriber C is
the supervisor. If C wants to join the conversation, it sends a request to A. If A permits, the
226
three-party conference can be held. In this example, C is called the active participant of the
conference, A is the voice mixer, and B is the original participant of the conversation.
Silent monitor and barge in services can be considered as the extensions of three-party conference.
To distinguish them with traditional three-party conference, these two services are called three-party
conference in active participation mode.
Calling party control
The calling party control service allows the called party to resume the conversation with the calling
party by picking up the phone within the specified time. For example, subscriber A is the calling party;
subscriber B is the called party. The on-hook delay is set to m seconds on the voice subscriber line of
subscriber B. After the call between A and B is established, if the calling party A hangs up first, the
call is ended up. If the called party B hangs up first, it can resume the call with A by picking up the
phone within m seconds. After that, no matter how many times B hangs up within m seconds, it can
resume the call with A by picking up the phone.
In this example, after B hangs up for the first time, A hears silent tones from the headphone within m
seconds. If subscriber C dials subscriber B during this time, the telephone of B will not ring, and C will
hear busy tones.
Door opening control
The door opening control service allows a user to open a door remotely. The process is as follows:
1.
User A who wants to enter a door calls user B.
2.
After the session is established, user B enters a password starting with an asterisk (*) and
ending with a pound (#) on the phone:
{
{
If the entered password is correct (the password matches the predefined door opening
control password for the voice subscriber line), the door control relay opens the door. After a
predefined door open duration, the door control relay will lock the door automatically.
If the entered password is incorrect, the door cannot be opened.
Support for SIP voice service of VCX
Together with a server, the VCX implements the application of multiple voice features such as Silent
Monitor, Camp On, and FwdMail Toggle by using the H3C proprietary SIP Feature messages. All
these voice features are called Feature service.
Hardware compatibility with call services
Call services are not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
227
•
MSR3600-51F.
Call services configuration task list
For the service features of call waiting, call forwarding, incoming call barring, and outgoing call
barring, the system supports the following configuration methods:
•
The system administrator performs configurations by using command lines on the device. This
method allows for more configuration options.
•
Terminal subscribers perform configurations on telephones for convenience of use and for
relieving the burden on the system administrator.
By default, the device supports the following functions, without any configuration:
•
Call backup function.
•
Functions (processing transfer messages) of the recipient gateway and final recipient gateway.
•
Functions (processing call hold/recover request messages) of the gateway on the held party
side.
•
Functions (processing forwarding messages) of the gateway that receives forwarding request
messages, namely, the originating gateway.
Task
Remarks
Configuring call waiting
Optional.
Configuring call hold
Optional.
Configuring call forwarding
Optional.
Configuring call transfer
Optional.
Configuring call backup
Optional.
Configuring hunt group
Optional.
Configuring incoming call barring
Optional.
Configuring outgoing call barring
Optional.
Configuring MWI
Optional.
Configuring three-party conference
Optional.
Configuring silent monitor and barge in
Optional.
Configuring calling party control
Optional.
Configuring door opening control
Optional.
Configuring Feature service
Optional.
Configuring a number priority peer
Optional.
Configuring call waiting
The device supports two call waiting configuration methods:
•
Subscribers perform configurations by using keys on a telephone terminal.
•
The system administrator performs configurations by using command lines on the device.
228
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling and disabling call waiting by using keys
The device allows you to configure call waiting for FXS voice subscriber lines through telephones.
The call waiting configuration made on a telephone is also valid for the device. For example, after
you enable call waiting on a telephone, you can view the operation result by using the display this
command in corresponding voice subscriber line view.
To enable and disable the call waiting feature:
Enable keys
Disable keys
*58#
#58#
Configuring call waiting by using command lines
Enabling call waiting
In addition to being enable or disable the call waiting feature, you can set parameters related to this
feature. These parameters include the number of times a call waiting tone pattern is played, the
number of tones in a call waiting tone pattern, and the interval for playing a call waiting tone pattern.
To enable call waiting:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber
line view.
subscriber-line line-number
N/A
3.
Enable call waiting.
call-waiting enable
Disabled by default.
Optional.
4.
Configure call waiting
tone parameters.
call-waiting { cwi-count
number | cwi-duration length |
cwi-interval length }
By default, a call waiting tone pattern is
played once, a call waiting tone pattern
contains two tones, and if the value of
cwi-count number is greater than 1 the
interval for playing a call waiting tone
pattern is 15 seconds.
Configuring a call waiting priority level
A priority level applies to the features of call waiting, call forwarding, and hunt group only.
By default, the priority levels for hunt group, call forwarding, and calling waiting are 1, 2, and 3
respectively. The smaller the value is, the higher the priority level is. When you change the priority
level of a feature, make sure that different features have different priority levels.
To configure a call waiting priority level:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
229
Step
3.
Configure a priority level for
call waiting.
Command
call-waiting priority level
Remarks
Optional.
The default is 3.
Configuration example
Enable call waiting for the voice subscriber line of Telephone A. Place a call from Telephone C to
Telephone A, which is in a conversation with Telephone B. The subscriber at Telephone A hears call
waiting tones, and Telephone C is not rejected but waits for Telephone A to answer the new call.
# Enable call waiting for the voice subscriber line of Telephone A.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-waiting enable
Configuring call hold
This section describes how to enable call hold and configure the tone playing mode.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling call hold using command lines
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable call hold.
call-hold enable
Disabled by default.
NOTE:
This command is applicable only for the FXS voice subscriber line.
Configuring the tone playing mode for call hold
There are two tone playing modes for call hold:
•
inactive—Silent mode. In this mode, the calling party does not play any tones to the called
party during call hold.
•
sendonly—Playing mode. In this mode, the calling party plays the specified tones to the called
party during call hold. Before specifying the playing mode for call hold, you need to configure
the media resource. For more information about media resource configuration, see
"Configuring customizable IVR."
If you specify sendonly as the tone playing mode for call hold, you can specify the tones played to
the called party using one of the following methods:
•
Specify the media resources ID. During call hold, the calling party plays the specified tones to
the called party. Before specifying the playing mode for call hold, you need to configure the
230
media resource. For information about media resource configuration, see "Configuring
customizable IVR."
•
Specify the MoH number. During the call hold, the third party (the music server) plays the tones
to the called party according to the MoH number. This method is available only when the device
operates as the SIP trunk device. For more information about SIP trunk, see "Configuring SIP
trunk." Before specifying the MoH number, make sure that the playing mode of the call hold
initiator is inactive (silent mode).
To configure tone playing mode of the call hold feature:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure the tone playing
mode for call hold.
call-hold-format { inactive |
sendonly [ media-play media-id ]
| moh-number string ] }
The default is inactive.
Configuration example
Enable call hold for the voice subscriber line of Telephone A. Telephone A and Telephone B are in a
conversation. The subscriber at Telephone A can interrupt the conversation with Telephone B by
pressing the hook flash, and place a call to Telephone C after hearing a dial tone. After the call with
Telephone C is established, the subscriber at Telephone A can switch between Telephone B and
Telephone C by pressing the hook flash and then pressing key 2. If the subscriber presses the hook
flash and then presses key 1, Telephone A will release the current call with Telephone C and resume
the hold call with Telephone B.
# Enable call hold for the voice subscriber line of Telephone A.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-hold enable
Configuring call forwarding
The device supports four different types of call forwarding, to cover all scenarios.
The device supports two call forwarding configuration methods:
•
Subscribers perform configurations by using keys on a telephone terminal.
•
The system administrator performs configurations by using command lines on the device.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling and disabling call forwarding by using keys
The device supports the call forwarding configuration for the voice subscriber line on telephones.
After you enable call forwarding on a telephone, you can view the corresponding operation result by
using the display this command in voice subscriber line view.
To enable and disable a type of call forwarding:
231
Enable keys
Disable keys
Remarks
*57*number#
#57#
Enable and disable call forwarding unconditional, where
"number" represents a forwarded-to number.
*40*number#
#40#
Enable and disable call forwarding busy, where "number"
represents a forwarded-to number.
*41*number#
#41#
Enable and disable call forwarding no reply, where
"number" represents a forwarded-to number.
*60*number#
#60#
Enable and disable call forwarding unavailable, where
"number" represents a forwarded-to number.
Configuring call forwarding by using command lines
In practice, you should set a reasonable, valid forwarded-to number and avoid setting the
forwarded-to number to a wrong number or the original called number.
Enabling call forwarding unconditional
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable call forwarding
unconditional.
call-forwarding unconditional
enable forward-number number
Disabled by default.
Enabling call forwarding no reply
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber
line view.
subscriber-line line-number
N/A
3.
Enable call forwarding
no reply.
call-forwarding no-reply enable
forward-number number
Disabled by default.
Enabling call forwarding busy
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable call forwarding busy.
call-forwarding on-busy enable
forward-number number
Disabled by default.
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable call forwarding
call-forwarding unavailable
Optional.
Enabling call forwarding unavailable
232
Step
unavailable.
Command
Remarks
enable forward-number number
Disabled by default.
Configuring call forwarding priority level
A priority level applies to only the features of call waiting, call forwarding, and hunt group.
By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3
respectively. The smaller the value is, the higher the priority level is. When you change the priority
level of a feature, make sure that different features have different priority levels.
To configure a call forwarding priority level:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure a call forwarding
priority level.
call-forwarding priority level
Optional.
The default is 2.
Configuration example
Call forwarding busy
Place a call from Telephone A to Telephone B. The system forwards the call to Telephone C when
Telephone B is busy. Finally, Telephone A and Telephone C start a conversation.
# Enable call forwarding busy for the voice subscriber line of Telephone B and forward the call from
Telephone A to Telephone C (3000).
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0] call-forwarding on-busy enable forward-number 3000
Call forwarding unconditional
Place a call from Telephone A to Telephone B. The system forwards the call to Telephone C
unconditionally. Finally, Telephone A and Telephone C start a conversation.
# Enable call forwarding unconditional for the voice subscriber line of Telephone B and forward the
call from Telephone A to Telephone C (3000).
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0] call-forwarding unconditional enable forward-number 3000
Call forwarding no reply
Place a call from Telephone A to Telephone B. The system forwards the call to Telephone C when
Telephone B fails to answer the call. Finally, Telephone A and Telephone C start a conversation.
# Enable call forwarding no reply for the voice subscriber line of Telephone B and forward the call
from Telephone A to Telephone C (3000).
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0] call-forwarding no-reply enable forward-number 3000
233
Call forwarding unavailable
Place a call from Telephone A to Telephone B. The system forwards the call to Telephone C when the
subscriber line of Telephone B is unavailable. Finally, Telephone A and Telephone C start a
conversation.
# Enable call forwarding unavailable for the voice subscriber line of Telephone B and forward the call
from Telephone A to Telephone C (3000).
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0] call-forwarding unavailable enable forward-number 3000
Configuring call transfer
This section describes the procedure for configuring call transfer by using command lines.
Configuration prerequisites
•
The router is equipped with an FXS voice interface card.
•
The call hold feature is enabled for the voice subscriber line of the call transfer originator.
Configuring call transfer by using command lines
Call transfer is dependent on call hold. Therefore, you must enable call hold before call transfer.
After the recipient receives a transfer request from the originator, no session will be established
between the recipient and the final recipient if the recipient does not support a Refer message, or the
final recipient is busy or fails to answer the call. However, the call between the originator and the
recipient can be re-established.
To enable call transfer and configure the call transfer start delay:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
Disabled by default.
3.
Enable call transfer.
call-transfer enable
4.
Configure a call transfer start
delay.
call-transfer start-delay number
This command applies only to the
originator gateway. For the
recipient and final recipient
gateways, the corresponding
functions are enabled.
Optional.
The default is 3 seconds.
Configuration example
Place a call from Telephone A to Telephone B to establish a conversation, then perform a hookflash
to put Telephone B on hold. Place a call from Telephone A to Telephone C, and then hangs up. Now
the conversation between Telephone B and Telephone C is established, and the call transfer by
Telephone A is completed.
# Enable call transfer for the voice subscriber line of Telephone A.
234
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0] call-transfer enable
Configuring call backup
By default, the call backup function is enabled on the device.
The system supports two types of call backup:
•
Strict call backup
One of the following three conditions will trigger strict call backup:
•
{
The device does not receive any reply from the peer after sending out a call request.
{
The device fails to initiate a call to the IP network side.
{
The device fails to register on the voice server.
Loose call backup
Loose call backup is triggered if any of the above mentioned three conditions or the following
condition happens: the device receives a reject reply (with a number from 3xx to 6xx except 300,
301, 302, 305, 401, 407, and 422) after sending a call request.
To configure the call backup mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Specify the loose call backup
mode.
Optional.
backup-rule loose
By default, strict call backup is
applied.
Optional.
4.
Set the time duration for call
backup.
Defaults to 5 seconds.
voip timer voip-to-pots time
For more information about the
voip timer voip-to-pots
command, see Voice Command
Reference.
Configuring hunt group
This section describes the procedures for enabling and configuring a hunt group.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling hunt group
To enable hunt group:
Step
Command
Remarks
1.
system-view
N/A
Enter system view.
235
Step
Command
Remarks
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable hunt group for the
voice subscriber line.
hunt-group enable
Disabled by default.
NOTE:
To use the hunt group feature, you need to configure the hunt-group enable command on all
involved voice subscriber lines.
Configuring hunt group priority level
A priority level applies to the features of call waiting, call forwarding, and hunt group only.
By default, the priority levels for hunt group, call forwarding (excluding call forwarding unconditional),
and call waiting are 1, 2, and 3 respectively. The smaller the value is, the higher the priority level is.
When you change the priority level of a feature, make sure that different features have different
priority levels.
To configure a hunt group priority level:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure a priority level for
hunt group.
hunt-group priority level
Optional.
1 by default.
NOTE:
The call forwarding unconditional feature is free from the restriction of priority level. If the call
forwarding unconditional feature is enabled, the other types of call forwarding, call waiting, and hunt
group features will never take effect.
Configuration example
Telephone B and Telephone C have the same number, but Telephone B has a higher priority than
Telephone C (the POTS entity priority of Telephone B is configured in POTS voice view using the
priority command). Place a call from Telephone A to Telephone B and Telephone B is busy. In this
case, the hunt group service enables Telephone C to have a conversation with Telephone A.
# Enter voice subscriber line view.
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0] hunt-group enable
[Sysname-subscriber-line2/0] quit
# Enable the hunt group feature for the voice subscriber lines.
[Sysname] subscriber-line 2/1
[Sysname-subscriber-line2/1] hunt-group enable
236
Configuring incoming call barring
When you do not want to receive any incoming call, you can enable incoming call barring (that is, the
Do Not Disturb feature).
The device supports two incoming call barring configuration methods:
•
Subscribers perform configurations by using keys on a telephone terminal.
•
The system administrator performs configurations by using command lines on the device.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling and disabling incoming call barring by using keys
The device supports the incoming call barring configuration for the FXS voice subscriber line on
telephone terminals. After you enable incoming call barring on a telephone, you can view the
corresponding operation result by using the display this command in voice subscriber line view.
Use the following keys to enable and disable incoming call barring:
Enable keys
Disable keys
*56#
#56#
Enabling incoming call barring by using command lines
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable incoming call barring.
dialin-restriction enable
Disabled by default.
Configuration example
Incoming call barring enabled on the voice subscriber line of Telephone A. When you place a call
from Telephone B to Telephone A, the line between Telephone A and Telephone B is directly cleared
and the subscriber at Telephone B hears busy tones.
# Enter voice subscriber line view.
<Sysname> system-view
[Sysname] subscriber-line 1/0
# Enable incoming call barring for the voice subscriber line.
[Sysname-subscriber-line1/0] dialin-restriction enable
237
Configuring outgoing call barring
When subscribers do not want others to use their telephones, they can set a password to lock their
telephones. Outgoing call barring can achieve this purpose. When they want to make calls, they can
disable outgoing call barring.
The device supports two outgoing call barring configuration methods:
•
Subscribers perform configurations by using keys on a telephone terminal.
•
The system administrator performs configurations by using command lines on the device.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling and disabling outgoing call barring by using keys
The device supports the outgoing call barring configuration for the FXS voice subscriber line on
telephone terminals. After you enable outgoing call barring on a telephone, you can view the
corresponding operation result by using the display this command in voice subscriber line view.
To enable and disable outgoing call barring:
Enable keys
Disable keys
Remarks
*54*number#
#54*number#
The string "number" represents a password. You
also need to enter the password when disabling
outgoing call barring.
Enabling outgoing call barring by using command lines
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable outgoing call barring.
dialout-restriction enable password
{ cipher | simple } password
Disabled by default.
Configuration example
Telephone A does not want anyone to use this telephone to make calls, and sets the password 1234.
When the subscriber at Telephone A calls Telephone B, the line between Telephone A and Telephone
B is directly cleared.
# Enter voice subscriber line view.
<Sysname> system-view
[Sysname] subscriber-line 1/0
# Enable outgoing call barring and set the password to 1234.
[Sysname-subscriber-line1/0] dialout-restriction enable password cipher 1234
238
Configuring MWI
This section describes configuration procedures for enabling, disabling, displaying, and maintaining
MWI.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling and disabling MWI
Configure MWI using command lines to enable or disable the feature, and to set the message
waiting tone duration.
To configure MWI:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable MWI.
mwi enable
Disabled by default.
4.
Configure the duration of
message waiting tone.
mwi tone-duration length
Optional.
The default is 2 seconds.
Specifying the voice mailbox server
The voice gateway sends a SUBSCRIBE to the server, and receives a NOTIFY from the server if the
subscription is successful, and gets the status of the voice mailbox afterwards.
To specify the voice mailbox server:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP client view.
sip
N/A
Specify the voice mailbox
server.
mwi-server { dns domain-name |
ipv4 ip-address } [ expires
seconds ] [ port port-number ]
[ retry seconds ] [ tcp | tls ]
[ scheme { sip | sips } ] { bind |
no-bind { loose | strict } }
Not specified by default.
4.
Displaying and maintaining MWI
Task
Command
Remarks
Display the information of MWI.
display voice ss mwi { all |
number number } [ | { begin |
exclude | include }
Available in any view.
239
Task
Command
Remarks
regular-expression ]
Display subscription information.
display voice sip
subscribe-state [ | { begin |
exclude | include }
regular-expression ]
Available in any view.
Configuring three-party conference
The device offers two three-party conference configuration methods:
•
Subscribers configure the function by pressing keys on a telephone.
•
The system administrator configures the function through CLI on the device.
Configuration prerequisites
•
Install an FXS voice card on the device
•
Install a voice processing module (VPM) on the main board of the device
•
Enable the call hold function in the voice subscriber line of the conference control device
Enabling three-party conference by using keys
The device supports configuring the three-party conference function of the FXS voice subscriber line
through a telephone.
The three-party conference configuration performed on a telephone also takes effect on the device.
For example, after you configure three-party conference through a telephone, and then execute the
display this command in the corresponding subscriber line, you can see the corresponding
operation results.
The three-party conference function depends on the call hold function. Therefore, you need to
enable the call hold function before configuring three-party conference.
Enabling the three-party conference service in voice subscriber line view will invalidate the local call
identification function (if configured). For more information about the configuration of the local call
identification function, see the distinguish-localtalk command in Voice Command Reference.
To enable three-party conference by using keys:
Enable keys
Disable keys
*33#
#33#
Enabling three-party conference by using command lines
The three-party conference function depends on the call hold function. Therefore, you need to
enable the call hold function before configuring three-party conference.
Configuring the three-party conference service in voice subscriber line view will invalidate the local
call identification function. For more information about the configuration of the local call identification
function, see the distinguish-localtalk command in Voice Command Reference.
To configure three-party conference by using command lines:
240
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable three-party
conference.
conference enable
Disabled by default.
Configuration example
Place a call from Telephone A to Telephone B and hold the call. Then, place a call from Telephone A
to Telephone C. After success, press the hook flash. After hearing the prompt tone, press 3 to
establish a three-party conference. At this time, subscribers of Telephones A, B, and C can have a
conversation simultaneously.
# Enable three-party conference for the voice subscriber line of Telephone A.
<Sysname> system-view
[Sysname] subscriber-line 2/0
[Sysname-subscriber-line2/0]call-hold enable
[Sysname-subscriber-line2/0] conference enable
Configuring silent monitor and barge in
You can configure silent monitor and barge in (that is, three-party conference in active participation
mode) either by using the keys on a telephone or by using the command lines on the device.
Configuration prerequisites
•
Install an FXS voice card on the device.
•
Install a VPM on the main board of the device.
•
The device has registered with the VCX so that all sessions will be established through the
VCX.
Configuring three-party conference in active participation
mode by using keys
The device supports configuring the three-party conference in active participation mode of the FXS
voice subscriber line through a telephone.
The configuration performed on the telephone also takes effect on the device. You can execute the
display this command on the corresponding subscriber line to check the configuration.
Configuring the three-party conference service in active participation mode on a telephone will
invalidate the local call identification function (if configured). For more information about the
configuration of the local call identification function, see the distinguish-localtalk command in Voice
Command Reference.
To configure three-party conference in active participation mode on a telephone:
Enable keys
Disable keys
*34#
#34#
241
Configuring three-party conference in active participation
mode by using command lines
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable three-party
conference in active
participation mode.
joined-conference enable
Disabled by default.
For the detailed configuration example, see "Silent monitor and barge in."
Configuring calling party control
This section describes the procedure for enabling and configuring calling party control.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Configuring calling party control
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Enable calling party control
and set the on-hook delay
time of the called party.
timer called-hookon-delay
seconds
Calling party control is disabled by
default, that is, the on-hook delay
of the called party is set to 0.
Configuring door opening control
This section describes the procedure for enabling and configuring door opening control.
Configuration prerequisites
Install an SIC audio card on the device on which the door opening-enabled FXS voice subscriber line
resides.
Configuring door opening control
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
subscriber-line line-number
N/A
242
Step
Command
Remarks
door-relay password [ simple |
cipher ] password [ time
seconds ]
Optional.
view.
3.
Enable the door opening
control service and set a
password for opening the
door and the door open
duration before the door
control relay locks the door.
Door opening control is disabled
by default.
Configuring Feature service
For the convenience of operation, VCX devices provide various features. Some of these features are
directly implemented through Feature messages and others are implemented through SIP
messages. All these features are implemented through exchange of special codes with VCX devices.
They are called Feature service.
After the system administrator enables the setting of the Feature service by using command lines,
you can dial some special codes on telephone terminals to implement these features. The setting of
the Feature service starts with an asterisk (*). After the setting is completed, you will hear the system
playing tones indicating whether the setting succeeds or fails.
Configuration prerequisites
The router is equipped with an FXS voice interface card.
Enabling and disabling Feature service setting by using keys
The device supports the Feature service configuration for the FXS voice subscriber line on telephone
terminals.
To enable and disable the setting of the Feature service:
Feature name
Silent Monitor
Enable keys
*425*destination#
Disable keys
None
Remarks
On the VCX, Telephone B has the right
to monitor Telephone A. When
Telephone A is in conversation,
Telephone B can dial the feature code to
monitor Telephone A, but Telephone A
does not know it is being monitored.
The feature is automatically disabled
upon on-hook.
Barge In
*428#
None
During monitoring, the supervisor can
dial the feature code to optionally join a
conversation and place a three-party
conference call.
The feature is automatically disabled
upon on-hook.
Transfer To
Voicemail
*441*destination#
None
A subscriber can dial the feature code
plus a destination mailbox number to
transfer incoming calls to the voice
mailbox. The destination mailbox
number must be an existing one.
Applied only once
243
Feature name
Park
Enable keys
*444*park_number
#
Disable keys
None
Remarks
In conversation, a subscriber can dial
the feature code to park the call to a
park extension designated by the
server. Within a certain period of time,
the subscriber can dial the park
extension on any telephone that is
registered with the server to retrieve the
call.
Applied only once
Do Not Disturb
Toggle
Directed Pickup
*446#
*455*pwd*pickup_
number#
*446#
None
A subscriber can dial the feature code to
enable or disable the DND feature. If the
feature is enabled, all incoming calls will
be rejected.
This feature allows a subscriber to
answer a call ringing on the phone of a
specific subscriber. To answer the call,
the subscriber enables the Directed
Pickup feature (feature code *455),
enters a security code, and then enters
the extension of the ringing phone. This
transfers the call to the subscriber.
Applied only once
Config Forward
Universal
*465*fwd_number
#
Config Forward Busy
*467*fwd_number
#
Config Forward Ring
No Answer
Config Remote Fwd
Universal
Hunt Group Login
Toggle
Retrieve Voice Mail
*466*fwd_number
#
*468*src_number*f
wd_number#
*971*hunt_group_
number#
*600 or *600*user#
*465#
A subscriber can dial the feature code to
enable the feature to redirect all
incoming calls to the forwarded-to
number unconditionally.
*467#
A subscriber can dial the feature code to
enable the feature to redirect all
incoming calls to the forwarded-to
number when the subscriber is busy.
*466#
A subscriber can dial the feature code to
enable the feature to redirect all
incoming calls to the forwarded-to
number when the subscriber does not
answer the call within a certain period of
time.
*468*src_number#
On the VCX, Telephone A is assigned
the right to enable call forwarding and
specify a forwarded-to number for
Telephone B. Thus, all incoming calls to
Telephone B will be redirected to the
forwarded-to number.
*971*hunt_group_
number#
After a hunt group is established on the
VCX voice server, a subscriber can dial
the feature code to join or leave the hunt
group. When a subscriber dials the hunt
group number, the idle telephone with
the highest priority in the hunt group will
first ring.
None
A subscriber can dial the feature code to
retrieve the voice mailbox. The
subscriber can perform operations
according to the prompts.
Applied only once.
FwdMail Toggle
*440#
*440#
244
A subscriber can dial the feature code to
Feature name
Enable keys
Disable keys
Remarks
enable or disable the feature. If the
feature is enabled, any new call will be
transferred to the voice mailbox after the
telephone rings only once.
Block Caller Id
Toggle
Block CallId for
Current Call
*889#
*890*destination
number#
*889#
None
A calling subscriber can dial the feature
code to hide or display the calling
number. If the feature is enabled, the
called subscriber cannot see the calling
number when a call is originated.
A subscriber can dial the feature code to
hide the calling number for the current
call and the called subscriber cannot
see the calling number.
Applied only once.
Subscriber Speed
Dial (range)
*601*code#
None
After a speed dial number is configured
for a subscriber on the VCX, the
subscriber can dial the feature code to
originate a call to the corresponding
telephone.
Applied only once.
Malicious Call Trace
*119#
None
After a subscriber dials the feature code,
the VCX sends X-ISDN tunnel
messages to the peer gateway for
processing.
Applied only once.
Block Barge In
Camp On
*429#
*469*destination_n
umber#
*429#
None
During a conversation, if the local dials
the feature code, no voice data will be
sent so that the supervisor cannot
monitor or barge in the conversation.
When the destination telephone is in a
conversation, the local end can dial the
feature code to ask the peer end
(destination telephone) to call back.
After the peer end hangs up, the local
end is alerted. The local end picks up
the phone and the peer end is alerted.
After the peer end picks up the phone, it
can communicate with the local end.
Applied only once.
Serial Calling
*471*destination_n
umber#
None
In a conversation, the local end dials the
feature code to transfer the call from the
peer end to the destination telephone.
After the destination telephone is on
hook, the local end is alerted and can
pick up the phone to resume the
conversation with the peer end. The
above process can be repeated to
perform serial call transfer.
Applied only once.
Configuring Feature service by using command lines
The feature service indicates the service that is used together with the VCX. When you need to
interact with the VCX by using telephone keys, you need to adopt out-of-band named telephone
245
event (NTE) transmission to send the DTMF digits to the VCX. The execution of the feature permit
command does not enable out-of-band NTE transmission, and you need to execute the outband nte
command on the called entity to enable it. For more information about out-of-band NTE transmission,
see "Configuring voice entities."
To configure the Feature service:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice subscriber line
view.
subscriber-line line-number
N/A
3.
Configure the feature
service.
feature { deny | permit }
deny by default.
Configuration example
The system is connected with a VCX. The subscriber at Telephone A wants to implement the Feature
service provided by the VCX.
# Enter voice subscriber line view.
<Sysname> system-view
[Sysname] subscriber-line 1/0
# Enable the setting of the Feature service for the voice subscriber line.
[Sysname-subscriber-line1/0] feature permit
Configuring a number priority peer
To configure a number priority peer:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
Optional.
4.
Configure the
number-priority peer.
number-priority peer enable
By default, a number starting with
an asterisk (*) or a pound sign (#)
will first match against a service
feature code.
After the number-priority peer enable command is configured, a dialed number will match first
against a voice entity match template and then a service feature code. For example, if a service
feature code is *40*1234 and the match template *40 is configured for a voice entity, *40*1234 dialed
by a subscriber will first match the number template *40 (*40 is dialed out as the called number), and
the feature corresponding to the service feature code *40*1234 will not be triggered.
Call services configuration examples
This section provides call services configuration examples.
246
Call waiting
Network requirements
As shown in Figure 71, place a call from Telephone C to Telephone A which is already engaged in a
call with Telephone B, and the call will not be rejected. Just like a normal call, the subscriber at
Telephone C will hear ringback tones, while the subscriber at Telephone A will hear call waiting tones,
as a reminder that another call is waiting on the line.
Figure 71 Network diagram
Router A
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure that Router A, Router B and Router C are
routable to each other.
1.
Configure Router A:
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterA-voice-dial-entity3000] match-template 3000
[RouterA-voice-dial-entity3000] quit
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] return
# Enable call waiting.
<RouterA> system-view
[RouterA] subscriber-line 1/0
[RouterA-subscriber-line1/0] call-waiting enable
2.
Configure Router B:
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 voip
247
[RouterB-voice-dial-entity1000] address sip ip 10.1.1.1
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
3.
Configure Router C:
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
[RouterC-voice-dial-entity3000] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip ip 10.1.1.1
[RouterB-voice-dial-entity1000] match-template 1000
Operation 1: When the subscriber at Telephone C dials 1000 to call Telephone A which is already
engaged in a call with Telephone B, the subscriber at Telephone C will hear ringback tones. At the
same time, the subscriber at Telephone A will hear call waiting tones, as a reminder that a call is
waiting on the line. If then the subscriber at Telephone A hangs up, the telephone will ring, and the
subscriber at Telephone A can pick up the phone to start a conversation with Telephone C.
Operation 2: When the subscriber at Telephone C dials 1000 to call Telephone A who is already
engaged in a call with Telephone B, the subscriber at Telephone A can press the flash hook to start a
conversation with Telephone C, and thus Telephone B is held. The subscriber at Telephone A can
press the flash hook again to continue the talk with Telephone B, and then Telephone C is held. Note
that, in this case, the call hold function must be enabled on the voice subscriber line connecting to
Telephone A.
Call forwarding busy
Network requirements
As shown in Figure 72, place a call from Telephone A to Telephone B. Router B forwards the call to
Telephone C when Telephone B is busy. Finally, Telephone A and Telephone C start a conversation.
Figure 72 Network diagram
Router A
1000
Telephone A
Router B
Eth1/1
10.1.1.1/24
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Eth1/1
20.1.1.1/24
Router C
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure that Router A, Router B and Router C are
routable to each other.
248
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
2.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 3000 voip
[RouterB-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterB-voice-dial-entity3000] match-template 3000
[RouterB-voice-dial-entity3000] quit
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] return
# Enable call forwarding busy.
<RouterB> system-view
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] call-forwarding on-busy enable forward-number 3000
3.
Configure Router C:
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
Call transfer
Network requirements
As shown in Figure 73, call transfer enables Telephone A to transfer Telephone B to Telephone C.
After the call transfer is completed, Telephone B and Telephone C are in a conversation.
The whole process is as follows:
1.
Call Telephone B from Telephone A, so that Telephone B and Telephone A are in a
conversation.
2.
Perform a hookflash at Telephone A to put the call with Telephone B on hold.
3.
Call Telephone C (3000) from Telephone A after hearing dial tones.
4.
Hang up Telephone A.
249
5.
Telephone B and Telephone C are in a conversation and call transfer is completed.
Figure 73 Network diagram
Router A
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure that Router A, Router B and Router C are
routable to each other.
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterA-voice-dial-entity3000] match-template 3000
[RouterA-voice-dial-entity3000] quit
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] return
# Enable call hold and call transfer.
<RouterA> system-view
[RouterA] subscriber-line 1/0
[RouterA-subscriber-line1/0] call-hold enable
[RouterA-subscriber-line1/0] call-transfer enable
2.
Configure voice entities on Router B.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
3.
Configure Router C.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
250
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
Hunt group
Network requirements
As shown in Figure 74, hunt group applies to the situation where multiple entities correspond to the
same number. When the voice subscriber line with the first highest priority is in use, the device
automatically connects an incoming call to the voice subscriber line with the second highest priority.
Telephone A1 (1000) and Telephone A2 (1000) are both connected to Router A, and Telephone A1
has a higher priority. Telephone B (2000) dials the number 1000. Because Telephone A1 has a
higher priority, Telephone B will be connected to Telephone A1. If Telephone C (3000) dials the
number 1000 after Telephone A1 and Telephone B are in a conversation, hunt group enables
Telephone C to have a conversation with Telephone A2.
Figure 74 Network diagram
Configuration procedure
Before performing the following configuration, make sure that Router A, Router B and Router C are
routable to each other.
1.
Configure Router A:
# Configure the voice entity with a higher priority.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] quit
# Configure the voice entity with a lower priority.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] line 1/1
[RouterA-voice-dial-entity1001] match-template 1000
[RouterA-voice-dial-entity1001] priority 4
# Enable hunt group for the voice subscriber lines.
251
[RouterA-voice-dial-entity1001] quit
[RouterA-voice-dial] quit
[RouterA-voice] quit
[RouterA] subscriber-line 1/0
[RouterA-subscriber-line1/0] hunt-group enable
[RouterA-subscriber-line1/0] quit
[RouterA] subscriber-line 1/1
[RouterA-subscriber-line1/1] hunt-group enable
2.
Configure voice entities on Router B.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip ip 10.1.1.1
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
3.
Configure voice entities on Router C.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 1000 voip
[RouterC-voice-dial-entity1000] address sip ip 20.1.1.1
[RouterC-voice-dial-entity1000] match-template 1000
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
MWI
Network requirements
As shown in Figure 75, Telephone A and Telephone B registered with the VCX through Router A and
Router B respectively. Configure a voice mailbox for Telephone A on the voice server, configure the
address and operation mode of the MWI server on Router A, and enable MWI on the voice
subscriber line of Telephone A.
Figure 75 Network diagram
Configuration procedure
1.
Configure VCX:
{
Configure call processing server
252
Open the Web interface of the server and select Central Management Console. Configure
the telephone information of Telephone A and Telephone B, with the subscriber passwords
as 1000 and 2000 respectively. Figure 76 uses Telephone A as an example.
Figure 76 Configuration page of call processing server (1)
When you enter the Edit Phone Profile page, as shown in Figure 77, type 9000 in the
Voice Mail Number field.
Figure 77 Configuration page of call processing server (2)
253
{
Configure unified messaging server
# Configure mailbox access number as 9000.
Open the Web interface of the server, select IP Messaging Web Provisioning to log in to
the unified messaging server, and click the Configuration link. You can see the
Configuration Option box, as shown in Figure 78.
Figure 78 Configuration page of unified messaging server
Select 9000 from the Main Voicemail Access Number List, as shown in Figure 79.
Figure 79 Access number configuration page
# Configure the voice mailbox of Telephone A
Click the Edit A Mailbox link, enter the mailbox number 1000 of Telephone A, and then
check that if the mailbox is created successfully. If you are prompted that the mailbox is not
present, select the Create/Delete Mailboxes link to create the mailbox of Telephone A, with
the mailbox number as 1000.
2.
Configure Router A:
# Configure voice entities.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip proxy
[RouterA-voice-dial-entity3000] match-template 9000
[RouterA-voice-dial-entity3000] quit
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] quit
[RouterA] quit
# Configure the subscriber line.
[RouterA] subscriber-line 1/0
254
[RouterA-subscriber-line1/0] mwi enable
[RouterA-subscriber-line1/0] quit
# Configure the SIP server.
[RouterA] voice-setup
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 100.1.1.101
[RouterA-voice-sip] mwi-server ipv4 100.1.1.101 bind
[RouterA-voice-sip] register-enable on
3.
Configure Router B:
# Configure voice entities.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 9000 voip
[RouterB-voice-dial-entity9000] address sip proxy
[RouterB-voice-dial-entity9000] match-template 9000
[RouterB-voice-dial-entity9000] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip proxy
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] quit
[RouterB-voice-dial] quit
[RouterB-voice] quit
# Configure the SIP server.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 100.1.1.101
[RouterB-voice-sip] register-enable on
After the above configuration, if a call is placed from Telephone B to Telephone A which is not picked
up within the ringing timeout interval, the call will be forwarded to the voice mailbox. Then, the
subscriber of Telephone B can leave a message and hang up. The server sends a NOTIFY indicating
that there is new message in the mailbox of Telephone A to the voice gateway. By picking up the
phone, the subscriber of Telephone A can hear the message waiting tone, and then dial the voice
mailbox access number and log in to the mailbox to get the message.
Three-party conference
Network requirements
As shown in Figure 80, place a call from Telephone A to Telephone B and after the call is established,
hold the call on Telephone B. Then, place a call from Telephone B to Telephone C. After success,
press the hook flash on Telephone B and press 3. Then a three-party conference can be established
among Telephones A, B and C.
255
Figure 80 Network diagram
Router A
Router B
Eth1/0
10.1.1.1/24
1000
Telephone A
Eth1/0
10.1.1.2/24
Eth1/0
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure that Router A, Router B and Router C are
routable to each other.
1.
Configure Router A:
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
# Enable the call hold function.
<RouterA> system-view
[RouterA] subscriber-line 1/0
[RouterA-subscriber-line1/0] call-hold
2.
enable
Configure Router B:
# Configure the voice entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 3000 voip
[RouterB-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterB-voice-dial-entity3000] match-template 3000
[RouterB-voice-dial-entity3000] quit
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip ip 10.1.1.1
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] return
# Enable the call hold function and the three-party conference function
256
<RouterB> system-view
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] call-hold
enable
[RouterB-subscriber-line1/0] conference enable
3.
Configure Router C:
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
[RouterC-voice-dial-entity3000] quit
[RouterC-voice-dial] entity 2000 voip
[RouterC-voice-dial-entity2000] address sip ip 20.1.1.1
[RouterC-voice-dial-entity2000] match-template 2000
[RouterC-voice-dial-entity2000] quit
# Enable the call hold function.
<RouterC> system-view
[RouterC] subscriber-line 1/0
[RouterC-subscriber-line1/0] call-hold enable
Now Telephone B, as the conference initiator, holds the three-party conference with the passive
participants Telephone A and Telephone C.
If you also enable three-party conference on the FXS lines of Telephone A and Telephone C on
Router A and Router C, then during the conference, a new call can be initiated from Telephone A or
Telephone C to invite another passive participant. In this way, conference chaining is implemented.
Silent monitor and barge in
Network requirements
•
Configure silent monitor for Telephone C to monitor the conversation between Telephone A and
Telephone B.
As shown in Figure 81, Telephone A and Telephone B are in a conversation. Dial the feature
code *425*Number of Telephone A# at Telephone C to monitor the conversation between
Telephone A and Telephone B.
•
Configure barge in for Telephone C to participate the conversation between Telephone A and
Telephone B.
Dial the feature code *428# at Telephone C to participate the conversation between Telephone
A and Telephone B.
257
Figure 81 Network diagram
Configuration procedure
1.
Configure the VCX:
Open the Web interface of the VCX and select Central Management Console. Configure the
information of Telephone A, Telephone B, and Telephone C. The following takes Telephone A
as an example.
Figure 82 Telephone configuration page
# Configure the monitoring authority
Click Features of number 1000 to enter the feature configuration page, and then click Edit
Feature of the Silent Monitor and Barge In feature to enter the page as shown in Figure 83.
258
Figure 83 Silent monitor and barge in feature configuration page (1)
Click Assign External Phones to specify that number 3000 has the authority to monitor
number 1000. After this configuration, the page as shown in Figure 84 appears.
Figure 84 Silent monitor and barge in feature configuration page (2)
After the above configuration, Telephone C with the number 3000 can monitor and barge in the
conversations of Telephone A with the number 1000.
2.
Configure Router A:
# Configure VoIP voice entities to Router B and Router C.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip proxy
[RouterA-voice-dial-entity3000] match-template 3000
[RouterA-voice-dial-entity3000] quit
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip proxy
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
# Configure a POTS voice entity, that is, configure the number of Telephone A as 1000.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] quit
[RouterA-voice-dial] quit
[RouterA-voice] quit
259
# Enable three-party conference in active participation mode.
[RouterA] subscriber-line 1/0
[RouterA-subscriber-line1/0] joined-conference enable
# Enable the setting of the Feature service.
[RouterA-subscriber-line1/0] feature permit
[RouterA-subscriber-line1/0] quit
# Specify the IP address 100.1.1.101 as the registrar and proxy server and enable the registrar.
[RouterA-voice] sip
[RouterA-voice-sip] registrar ipv4 100.1.1.101
[RouterA-voice-sip] proxy ipv4 100.1.1.101
[RouterA-voice-sip] register-enable on
3.
Configure Router B:
# Configure VoIP voice entities to Router A and Router C.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip proxy
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
[RouterB-voice-dial] entity 3000 voip
[RouterB-voice-dial-entity3000] address sip proxy
[RouterB-voice-dial-entity3000] match-template 3000
[RouterB-voice-dial-entity3000] quit
# Configure a POTS voice entity, that is, configure the number of Telephone B as 2000.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] quit
[RouterB-voice-dial] quit
# Specify the IP address 100.1.1.101 as the registrar and proxy server and enable the registrar.
[RouterB-voice] sip
[RouterB-voice-sip] registrar ipv4 100.1.1.101
[RouterB-voice-sip] proxy ipv4 100.1.1.101
[RouterB-voice-sip] register-enable on
4.
Configure Router C:
# Configure VoIP voice entities to Router A and Router B.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 1000 voip
[RouterC-voice-dial-entity1000] address sip proxy
[RouterC-voice-dial-entity1000] match-template 1000
# Configure the out-of-band NTE DTMF transmission for VoIP voice entity 1000.
[RouterC-voice-dial-entity1000] outband nte
[RouterC-voice-dial-entity1000] quit
[RouterC-voice-dial] entity 2000 voip
[RouterC-voice-dial-entity2000] address sip proxy
260
[RouterC-voice-dial-entity2000] match-template 2000
[RouterC-voice-dial-entity2000] quit
# Configure POTS voice entity, that is, configure the number of Telephone C as 3000.
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
[RouterC-voice-dial-entity3000] quit
[RouterC-voice-dial] quit
# Specify the IP address 100.1.1.101 as the registrar and proxy server and enable the registrar.
[RouterC-voice] sip
[RouterC-voice-sip] registrar ipv4 100.1.1.101
[RouterC-voice-sip] proxy ipv4 100.1.1.101
[RouterC-voice-sip] register-enable on
[RouterC-voice-sip] quit
[RouterC-voice] quit
# Enable the setting of feature service.
[RouterC] subscriber-line 1/0
[RouterC-subscriber-line1/0] feature permit
After the above configuration, dial feature code *425*1000# at Telephone C, and you can monitor the
conversation between Telephone A and Telephone C. If you want to participate in the conversation,
dial *428# at Telephone C.
261
Configuring call watch
The call watch function is only applicable to voice E1/T1 interfaces. The E1/T1 interfaces mentioned
in this document are all voice interfaces.
Overview
Call watch enables a voice device to decide whether an E1/T1 interface is available for setting up
calls for a callee by monitoring the state of the local interface or the IP connectivity to the remote
interface connected to the callee.
Figure 85 Network diagram
Voice device
PBX
E1/T1
User A
Backup link
Local link
Backup
network
VoIP network
User B
As shown in Figure 85, a call watch-enabled voice device is connected through a local link to user B
and through an E1 or T1 link to the PBX to which user A is attached. This provides VoIP
communication between the two users in normal cases. When the local link is detected as
unavailable, the voice device sets the E1/T1 interface to watch-out state, disabling the interface to
respond to calls from the PBX. Detecting that the E1/T1 link is unavailable, the PBX switches to the
backup link to forward calls from user A to user B. When the local link recovers, the watch-out state of
the E1/T1 interface is removed and the PBX can thus dial through the E1/T1 link. As a result, calls
from user A to user B are set up across the local link.
Call watch concepts
Call watch group
The call watch function is implemented through call watch groups, each monitoring the state of one
or multiple local links.
Monitoring rule
A call watch group is a set of monitoring rules each defining a local interface or remote IP address
monitored by the group.
A call watch group can monitor either local interfaces or remote IP addresses (that is, IP connectivity
to remote interfaces), but not both.
The state of the E1/T1 interface associated with a call watch group is set as follows:
262
•
If local interfaces are monitored, the E1/T1 interface is set to watch-out state when all the
monitored local interfaces are down.
•
If IP connectivity to remote interfaces is monitored, the E1/T1 interface is set to watch-out state
when all the monitored remote IP addresses are unreachable.
Call watch mode
A call watch group can operate in hard or soft mode on an E1/T1 interface.
•
In soft mode, the E1/T1 interface will be set to watch-out state after all the monitored links are
detected unavailable only if no calls are present on the interface.
•
In hard mode, the E1/T1 interface is set to watch-out state immediately after all the monitored
links are detected unavailable regardless of whether calls are present on the interface.
Hardware compatibility with call watch
Call watch is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
Configuring call watch for an E1/T1 interface
Configuring call watch for an E1/T1 interface includes these tasks:
•
Configuring a call watch group
•
Associating the E1/T1 interface with the call watch group
Configuring a call watch group
Configuring a call watch group creates the call watch rules in the group. To monitor the IP
connectivity to remote interfaces, a call watch group must collaborate with the NQA and Track
modules. Thus, configure the remote IP addresses to be monitored with NQA, and configure the
Track function to work with NQA to send probe messages to the call watch group. In a call watch
group, the rule for monitoring a remote interface is the track object ID associated with the interface.
You can create up to 255 call watch groups on each voice device.
You can configure a call watch group to monitor up to 16 local interfaces or remote IP addresses.
For more information about configuring the NQA function, see Network Management and Monitoring
Configuration Guide.
For more information about configuring the Track function, see High Availability Configuration Guide.
To configure a call watch group:
263
Step
Command
Remarks
1.
system-view
N/A
Enter system view.
•
2.
Create a rule in a call
watch group.
•
Specify a local interface in the rule
call-watch rule watch-number local
interface interface-type
interface-number
Specify a track object ID in the rule
call-watch rule watch-number
remote track track-entry-number
Use either command.
To create multiple rules in the
call watch group, repeat the
step.
Associating the E1/T1 interface with the call watch group
To adapt the state of an E1/T1 interface to the state of the monitored local links in a call watch group,
you must associate the E1/T1 interface with the call watch group. Note that, each E1/T1 interface
can be associated with only one monitor group.
To associate the E1/T1 interface with the call watch group:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1/T1 interface view.
controller controller-type number
N/A
3.
Associate the interface with the
call watch group.
•
•
In hard mode:
call-watch group watch-number hard
In soft mode:
call-watch group watch-number soft
Use either
command.
Displaying and maintaining call watch
Task
Command
Remarks
Display information about the call
watch groups associated with E1/T1
interfaces.
display call-watch status [ controller
controller-type controller-number ] [ |
{ begin | exclude | include }
regular-expression ]
Available in any
view.
Call-watch configuration examples
This section provides call-watch configuration examples.
Monitoring local interfaces
Network requirements
As shown in Figure 86, device Voice A is connected through two local links to Router B and through
an E1 link to the PBX to which Router A is connected to provide VoIP communication between
Router A and Router B in normal cases.
To enable device Voice A to disable interface E1 1/0 from routing calls destined for Router B when
interfaces Ethernet 1/1 and Ethernet 1/2 go down so that the PBX can immediately switch to a
backup link, do the following:
•
Configure a call watch group to monitor interfaces Ethernet 1/1 and Ethernet 1/2.
264
•
Apply the call watch group to interface E1 1/0 and configure the call watch group to work in hard
mode.
Figure 86 Network diagram
Configuration procedure
1.
Configure device Voice A:
# Configure an IP address for each interface. (Details not shown.)
# Configure call watch group 1 to monitor local interfaces Ethernet 1/1 and Ethernet 1/2.
<VoiceA> system-view
[VoiceA] call-watch rule 1 local interface ethernet 1/1
[VoiceA] call-watch rule 1 local interface ethernet 1/2
# Associate E1 1/0 with call watch group 1 in hard mode.
[VoiceA] controller e1 1/0
[VoiceA-E1 1/0] call-watch group 1 hard
2.
Configure Router A, Router B and device Voice B: configure an IP address for each interface.
(Details not shown.)
Monitoring remote IP addresses
Network requirements
As shown in Figure 87, device Voice A is connected through two local links to Router B and through
an E1 link to the PBX to which Router A is connected to provide VoIP communication between
Router A and Router B in normal cases.
To enable device Voice A to disable interface E1 1/0 from routing calls destined for Router B when
remote interfaces Ethernet 1/1 and Ethernet 1/2 on Router B are not IP reachable so that the PBX
can immediately switch to a backup link, do the following:
•
Configure a call watch group to work together with the NQA and Track module to monitor IP
connectivity to remote interfaces Ethernet 1/1 and Ethernet 1/2 on Router B.
•
Apply the call watch group to interface E1 1/0 and configure the call watch group to work in soft
mode.
265
Figure 87 Network diagram
Configuration procedure
1.
Configure device Voice A:
# Enable NQA server, configure two NQA test groups to monitor remote IP addresses 10.1.1.2
and 10.1.2.2, and associate the NQA test groups each with a track object.
<VoiceA> system-view
[VoiceA] nqa server enable
[VoiceA] nqa entry admin test1
[VoiceA-nqa-admin-test1] type icmp-echo
[VoiceA-nqa-admin-test1-icmp-echo] destination ip 10.1.1.2
[VoiceA-nqa-admin-test1-icmp-echo] frequency 1000
[VoiceA-nqa-admin-test1-icmp-echo] probe timeout 1000
[VoiceA-nqa-admin-test1-icmp-echo] reaction 1 checked-element probe-fail
threshold-type consecutive 1 action-type trigger-only
[VoiceA-nqa-admin-test1-icmp-echo] quit
[VoiceA] track 1 nqa entry admin test1 reaction 1
[VoiceA] nqa schedule admin test1 start-time now lifetime forever
[VoiceA] nqa entry admin test2
[VoiceA-nqa-admin-test2] type icmp-echo
[VoiceA-nqa-admin-test2-icmp-echo] destination ip 10.1.2.2
[VoiceA-nqa-admin-test2-icmp-echo] frequency 1000
[VoiceA-nqa-admin-test2-icmp-echo] probe timeout 1000
[VoiceA-nqa-admin-test2-icmp-echo] reaction 1 checked-element probe-fail
threshold-type consecutive 1 action-type trigger-only
[VoiceA-nqa-admin-test2-icmp-echo] quit
[VoiceA] track 2 nqa entry admin test2 reaction 1
[VoiceA] nqa schedule admin test2 start-time now lifetime forever
# Create call watch group 1, referencing the track object IDs.
[VoiceA] call-watch rule 1 remote track 1
[VoiceA] call-watch rule 1 remote track 2
# Associate interface E1 1/0 with call watch group 1 in soft mode.
[VoiceA] controller e1 1/0
[VoiceA-E1 1/0] call-watch group 1 soft
266
2.
Configure an IP address for each interface on Router A, Router B and Voice B. (Details not
shown.)
267
Configuring fax over IP
Traditional fax machines transmit and receive faxes over PSTN. Fax has gained wide acceptance
due to its many advantages, such as high transmission speed and simple operations. By far, G3 fax
machines dominant fax communications. A G3 fax machine adopts the signal digitizing technology.
Image signals are digitized and compressed internally, then converted into analog signals through a
modem, and finally transmitted into the PSTN switch through common subscriber lines.
FoIP means sending and receiving faxes over the Internet. Routers can provide the FoIP function
after the FoIP feature is added, on the basis of the VoIP function. Because the FoIP is the
Internet-based fax service, sending national and international faxes costs less.
The network diagram for FoIP is similar to that for VoIP. You just replace the IP phone with a fax
machine to implement the fax function. As long as you can use IP phones, you can use the fax
function. This makes the fax function very simple.
The following figure illustrates an FoIP system structure.
Figure 88 FoIP system structure
PSTN
Internet
Fax
PSTN
Fax
FoIP protocols and standards
IP real-time fax complies with the ITU-T T.30 and T.4 protocols on the PSTN side and the H.323 and
T.38 protocols on the IP network side.
•
T.30 protocol pertains to file and fax transmission over PSTN. It describes and regulates the
communication traffic of G3 fax machines over common telephone networks, signal format,
control signaling, and error correction to the full extent.
•
T.4 protocol is a standard protocol involving the G3 fax terminals for file transmission. It
provides a standard regulation for the G3 fax terminals on image encoding/decoding scheme,
signal modulation and speed, transmission duration, error correction, and file transmission
mode.
•
T.38 protocol pertains to the real-time G3 fax over IP networks. It describes and regulates the
communication mode, packet format, error correction and some communication flows of
real-time G3 fax over IP networks.
Fax flow
In FoIP, the call setup, handshake, rate training, packet transfer, and call release are always
real-time. From the perspective of users, FoIP is no different than faxing over PSTN.
Signals that a G3 fax machine receives and sends are modulated analog signals. Therefore the
router processes fax signals in a different way than it processes telephone signals. The router needs
to perform A/D or D/A conversion for fax signals (that is, the router demodulates analog signals from
PSTN into digital signals, or modulates digital signals from the IP network into analog signals), but
does not need to compress fax signals.
268
A real-time fax process consists of five phases:
1.
Fax call setup phase. This phase is similar to the process of a telephone call setup. The
difference is that the fax tones identifying the sending/receiving terminals are included.
2.
Prior-messaging phase. During this phase, fax faculty negotiation and training are performed.
3.
Messaging phase. During this phase, fax packets are transmitted in accordance with the T.4
procedure, and packet transmission is controlled (including packets synchronization, error
detection and correction, and line monitoring).
4.
Post-messaging phase. During this phase, control operations such as packet authentication,
messaging completion, and multi-page continuous transmission are performed.
5.
Fax call release phase. During this phase, the fax call is released.
Hardware compatibility with FoIP
FoIP is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
•
MSR3600-51F.
FoIP configuration task list
Before configuring FoIP, configure POTS and VoIP voice entities. For more information about the
configuration procedure, see "Configuring voice entities."
After VoIP configuration, you can make IP phone calls. Usually, the default FoIP configuration can be
used to send and receive faxes so long as a fax machine is connected. FoIP configuration is mainly
to set the specific FoIP parameters, or used for some particular situations where the fax cannot be
made by using the default transmit energy level of a gateway carrier.
Task
Remarks
Configuring fax interworking protocol
Required.
Enabling CNG fax switchover
Optional.
Enabling ECM for fax
Optional.
Configuring fax faculty transmission mode
Optional.
Configuring maximum fax rate
Optional.
Configuring fax training mode
Optional.
Configuring threshold of local training
Optional.
Configuring transmit energy level of gateway carrier
Optional.
Configuring T.38 faculty description compatibility
Optional.
Configuring global default parameters for fax
Optional.
269
Configuring fax interworking protocol
The device supports two fax protocols: T.38 protocol and standard T.38 protocol. The standard T.38
protocol should be selected for interworking with leading fax terminals in the industry. Since most
leading fax terminals in the industry do not support the local training mode, the end-to-end training
mode must be selected for interworking with them.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure a fax interworking protocol:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots | voip }
N/A
Optional.
By default, the number of two kinds
of redundant packets is 0.
•
5.
Configure a fax
interworking protocol.
•
Configure the T.38 fax
protocol:
fax protocol t38
[ hb-redundancy number |
lb-redundancy number ]
Configure the standard T.38
(UDP) fax protocol:
fax protocol standard-t38
[ hb-redundancy number |
lb-redundancy number ]
If the call control protocol is SIP, the
two commands can be used only for
the originator of the fax request
(using private T.38, standard T.38,
or fax pass-through protocol). When
a fax request is originated using
private T.38, standard T.38, or fax
pass-through protocol, the fax type
is decided according to the
configurations. The receiver of the
fax request responds to the
originator based on the type of the
fax request, and then establishes a
fax call.
Configuring the pass-through mode
The fax pass-through technology was developed primarily for the purpose of compressing and
transmitting T.30 fax packets that cannot be demodulated through packet switched networks. With
this technology, the devices on two sides can directly communicate over a transparent IP link, and
the voice gateways (routers) do not distinguish fax calls from voice calls. After detecting a fax tone in
an established VoIP call, the voice gateway checks whether the voice codec protocol is G.711. If not,
the voice gateway switches the codec to G.711. Then fax data is transmitted as voice data in the
pass-through mode.
In the pass-through mode, fax information is in the format of uncompressed G.711 codes, is
encapsulated in RTP packets between gateways, and occupies a fixed bandwidth of 64 Kbps.
Although the packet redundancy mechanism can reduce the packet loss ratio, the pass-through
mode is subject to factors such as packet loss ratio, jitter, and delay. Therefore, it is necessary to
ensure synchronization of the clocks on both sides. Fax pass-through is called VBD by ITU-T. That is,
fax or modem signals are transmitted over a voice channel using a proper coding method. So far, the
codecs supported are only G.711 A-law and G.711 µ-law. In addition, when the fax pass-through
function is enabled, the VAD function must be disabled to avoid fax failures.
270
You can implement the fax pass-through function on the voice gateway (router) in the following ways:
•
Configure the fax to work in the pass-through mode on both sides.
•
Negotiate the codec as G.711 and set the fax rate to disable on both sides. Then, disable the
VAD function to avoid fax failures. This method is used for the voice gateway to interwork with
other devices in the pass-through mode.
To configure the pass-through mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots |
voip }
N/A
Optional.
Pass-through mode is disabled for fax
by default.
5.
Configure the fax
pass-through mode.
fax protocol pcm { g711alaw |
g711ulaw }
If the call control protocol is SIP, this
command can be used only for the
originator of the fax request (using
private T.38, standard T.38, or fax
pass-through protocol). When a fax
request is originated using private
T.38, standard T.38, or fax
pass-through protocol, the fax type is
decided according to the
configurations. The receiver of the fax
request responds to the originator
based on the type of the fax request,
and then establishes a fax call.
Configuring the SIP modem pass-through function
The SIP modem pass-through function is mainly used for remote device management. Since the
VoIP network has replaced part of the traditional PSTN, VoIP devices are required to support the
modem pass-through function, which can help remote PSTN users to log in to internal network
devices through dialup.
To configure the SIP Modem pass-through function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial
program view.
dial-program
N/A
4.
Enter voice entity
view.
entity entity-number { pots | voip }
N/A
5.
Configure the codec
type and switching
mode for SIP modem
pass-through
function.
modem protocol pcm { standard |
nte-compatible } { g711alaw | g711ulaw }
Configure the NTE
payload type for the
modem compatible-param payload-type
6.
Optional.
271
By default, the SIP modem
pass-through function is not
configured.
Optional.
By default, the value of the
Step
Command
Remarks
NTE
compatible-switching
mode.
NTE payload type is 100.
This command is valid only
for the NTE-compatible
switching mode.
Enabling CNG fax switchover
Configuration prerequisites
•
VoIP configuration is completed, IP calls can be made successfully, and fax machines are
connected correctly.
•
The configuration on the XE 7000 is completed and the fax mailbox function is enabled.
Configuration procedure
The calling tone (CNG) fax switchover is used to implement the fax mailbox service through
communication with the XE 7000 device. When the local fax machine A originates a fax call to the
peer fax machine B, if B is busy or is unattended, A can send the fax call to the fax mailbox of the XE
7000. With CNG fax switchover enabled, the voice gateway can switch to the fax mode once it
receives a CNG from A.
To enable CNG fax switchover:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial
program view.
dial-program
N/A
4.
Enter voice entity
view.
entity entity-number { pots | voip }
N/A
5.
Enable CNG fax
switchover.
fax cng-switch enable
Disabled by default.
Enabling ECM for fax
As defined in ITU-T, the error correction mode (ECM) is required by the half-duplex and
half-modulation system running ITU-T V.34 protocol for fax message transmission. Besides, the G3
fax terminals working in full duplex mode are required to support half-duplex mode, that is, ECM.
The fax machines using ECM can correct errors, provide the automatic repeat request (ARQ)
function, and transmit fax packets in the format of HDLC frames. Fax machines using non-ECM
cannot correct errors and they transmit fax packets in the format of binary strings.
To use ECM fax machines on both sides, the gateway must support ECM.
Enable ECM mode for the POTS and VoIP entities corresponding to the fax sender and receiver in
the ECM mode.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure ECM for fax:
272
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots | vofr |
voip }
N/A
By default, ECM is disabled on the
gateway.
5.
Enable ECM for fax.
fax ecm
The configuration of this command
in voice entity view is invalid for the
FRF.11 trunk mode.
Configuring fax faculty transmission mode
In common fax applications, the participating fax terminals negotiate with the standard faculty (such
as V.17 and V.29 rate) by default. It means that they do not send each other non-standard facilities
(NSF) message frames. In some cases such as encrypted fax, both fax terminals adopt a (NSF to
negotiate. At the start of negotiation, both terminals first exchange NSF message frames, and then
negotiate the subsequent fax faculty for communication. NSF messages are standard T.30
messages and carry private information.
In order to use a nonstandard faculty for negotiation, the following conditions must be met:
•
Fax terminals must support nonstandard transmission mode.
•
The transmission mode must be set to a nonstandard mode in the POTS and VoIP entities for
both fax terminals.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure fax faculty transmission mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots |
voip }
N/A
5.
Configure the signal
transmission mode of fax
faculty.
fax nsf-on
By default, a standard faculty mode
is adopted for fax faculty
transmission.
Configuring maximum fax rate
You can configure the maximum fax rate according to the fax protocols. If the baud rate is set to a
value other than disable and voice, the configured value is adopted as the allowed maximum fax
rate.
273
If voice mode is adopted, the allowed maximum fax rate is determined first in accordance with voice
coding/decoding protocols.
•
If G.711 is used, the fax rate is 14,400 bps and the corresponding fax protocol is V.17.
•
If G.723.1 Annex A is used, the fax rate is 4800 bps and the corresponding fax protocol is V.27.
•
If G.726 is adopted, the fax rate is 14,400 bps and the corresponding fax protocol is V.17.
•
If G.729 is used, the fax rate is 7200 bps and the corresponding fax protocol is V.29.
If the fax rate is set to "disable", the fax function is disabled.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure the fax rate:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots |
voip }
N/A
By default, the voice mode is first
used to determine the fax rate.
5.
Configure the allowed
maximum fax rate.
fax baudrate { 2400 | 4800 | 9600
| 14400 | disable | voice }
Note that when the disable
keyword is provided, if the call
control protocol is SIP, private
T.38 and standard T.38 faxes are
disabled.
Configuring fax training mode
There are two fax training modes: local training and point-to-point training.
•
Local training—The gateways participate in the rate training between fax terminals. In the
local training mode, rate training is performed between fax terminals and gateways respectively,
and then the receiving gateway sends the training result of the receiving fax terminal to the
transmitting gateway. The transmitting gateway finalizes the packet transmission rate by
comparing the received training result with its own training result.
•
Point-to-point training—The gateways do not participate in the rate training between two fax
terminals. In this mode, rate training is performed between two fax terminals and is transparent
to the gateways.
Perform the following configuration in voice entity view.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure the fax training mode:
274
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots | vofr
| voip }
N/A
5.
Configure a fax training
mode.
fax train-mode { local | ppp }
By default, the PPP training is
adopted.
NOTE:
VoFR entities only support the PPP training mode.
Configuring threshold of local training
When rate training is carried on between fax terminals, the transmitting terminal transmits
"zero-filled" TCF data (the filling time per packet is 1.5±10% seconds) to the receiving fax terminal,
and the receiving fax terminal decides whether the current rate is acceptable according to the
received TCF data.
When the percentage of all-ones or all-zeros TCF data to the total number of TCP data is less than
the local training threshold, the current rate training succeeds. Otherwise, the current rate training
fails and you need to drop the rate for a local training operation again.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure the threshold percentage of local training:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots | vofr
| voip }
N/A
5.
Configure the fax training
mode.
fax train-mode local
By default, the PPP training is
adopted.
6.
Configure the threshold of
local training.
fax local-train threshold
threshold
By default, the threshold is 10.
NOTE:
When the local training mode is adopted, use the fax local-train threshold command to configure
the threshold in percentage. When the PPP training mode is adopted, the gateway does not
participate in rate training and the threshold of local training is not applicable.
275
Configuring transmit energy level of gateway carrier
Usually, the default transmit energy level of the gateway carrier is acceptable. If the fax cannot be set
up yet on the premise that other configurations are correct, you can attempt to adjust the transmit
energy level of the gateway carrier (that is, transmit energy level attenuation). A greater level
indicates greater energy. A smaller level indicates greater attenuation.
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure the transmit energy level of the gateway carrier:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Enter voice entity view.
entity entity-number { pots |
vofr | voip }
N/A
5.
Configure the transmit
energy level of the gateway
carrier.
fax level level
By default, the transmit energy level
of the gateway carrier is –15 dBm.
Configuring T.38 faculty description compatibility
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure the T.38 faculty description compatibility:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure the faculty set of the
voice gateway in H.323 slow
connection to contain T.38
faculty description..
voip h323-conf tcs-t38
By default, the faculty set contains
T.38 faculty description.
NOTE:
Because NetMeeting does not support T.38 faculty description parsing, you must disable the voice
gateway in H.323 slow connection mode from containing the T.38 faculty description in its faculty set
to interwork with NetMeeting.
276
Configuring global default parameters for fax
Configuration prerequisites
VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected
correctly.
Configuration procedure
To configure global default parameters for fax:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Configure the transmit
energy level of the gateway
carrier globally.
Optional.
default entity fax level level
By default, the transmit energy
level of the gateway carrier is –15
dBm.
Optional.
By default, T.38 (namely, T.38)
protocol is used for fax. The
number of low speed and high
speed redundant packets is 0.
5.
Configure the protocol for
interworking with other
devices globally.
default entity fax protocol { t38 |
standard-t38 } [ lb-redundancy
number | hb-redundancy
number ]
default entity fax protocol pcm
{ g711alaw | g711ulaw }
If the call control protocol is SIP,
this command can be used only
for the originator of the fax request
(using private T.38, standard
T.38, or fax pass-through
protocol). When a fax request is
originated using private T.38,
standard T.38, or fax
pass-through protocol, the fax
type is decided according to the
configurations. The receiver of the
fax request responds to the
originator based on the type of the
fax request, and then establishes
a fax call.
Optional.
6.
Configure the maximum fax
rate globally.
default entity fax baudrate
{ 2400 | 4800 | 9600 | 14400 |
disable | voice }
7.
Configure the fax negotiation
faculty globally.
default entity fax nsf-on
8.
Configure the fax training
mode globally.
default entity fax train-mode
{ local | ppp }
9.
Configure the threshold
default entity fax local-train
By default, the fax rate is
determined by the voice mode.
Note that when the disable
keyword is provided, if the call
control protocol is SIP, forwarding
of private T.38 and standard T.38
faxes will be disabled.
Optional.
277
By default, the fax negotiation is
based on the standard faculty.
Optional.
By default, the ppp training is
adopted.
Optional.
Step
percentage of local training
globally.
Command
Remarks
threshold threshold
By default, the threshold is 10.
You must carry out the default
entity fax train-mode local
command before the
configuration made by the default
entity fax local-train threshold
threshold command takes effect.
Optional.
10. Configure the ECM mode
globally.
default entity fax ecm
By default, the fax does not use
ECM.
11. Enable CNG fax switchover
globally.
default entity fax cng-switch
enable
Optional.
12. Configure the codec type
and switching mode for SIP
Modem pass-through
function globally.
default entity modem protocol
pcm { standard |
nte-compatible } { g711alaw |
g711ulaw }
Optional.
Disabled by default.
By default, the SIP Modem
pass-through function is not
configured.
For information about how to use the global default parameters for fax, see "Configuring voice
entities."
Displaying and maintaining FoIP configuration
Task
Command
Remarks
Display the statistics of the FoIP
module.
display voice fax statistics [ |
{ begin | exclude | include }
regular-expression ]
Available in any view.
Clear the statistics of the FoIP
module.
reset voice fax statistics
Available in user view.
FoIP configuration examples
This section provides FoIP configuration examples.
Configuring FoIP
Network requirements
As shown in Figure 89, the headquarters of a company in City B and its branch in City A need to
transmit and receive faxes over an IP network.
•
The IP addresses 1.1.1.1/24 and 2.2.2.2/24 are respectively assigned to the interfaces through
which the routers in City A and City B access the Internet.
•
Router A in City A and Router B in City B are connected to fax terminals through an FXS voice
subscriber line.
•
At the branch in City A, the number "0101001" is attached to the FXS voice subscriber-line
connected to the fax machine. At the headquarters in City B, the number "07552001" is
attached to the FXS voice subscriber-line connected to the fax machine. Standard T.38 (UDP)
protocol is used for fax communication.
278
Figure 89 Network diagram
010-1001
Router A
Line 1/1
Fax
0755-2001
Router B
1.1.1.1/24
2.2.2.2/24
Internet
Line 1/1
City A
Fax
City B
Configuration procedure
1.
Configure Router A:
# Configure the standard T.38 (UDP) fax protocol.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] default entity fax protocol standard-t38
# Configure VoIP voice entity 0755, and configure the IP address and the fax number of the
peer VoIP gateway as 2.2.2.2 and 0755.… respectively.
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
[RouterA-voice-dial-entity755] quit
# Specify 0101001 as the local fax number of POTS voice entity 1001.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/1
2.
Configure Router B:
# Configure the standard T.38 (UDP) fax protocol.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] default entity fax protocol standard-t38
# Configure VoIP voice entity 010, and configure the IP address and the fax number of the peer
VoIP gateway as 1.1.1.1 and 010.… respectively.
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Specify 07552001 as the local fax number of POTS voice entity 2001.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/1
Configuring SIP modem pass-through
Network requirements
As shown in Figure 90, enable SIP modem pass-through on Router A and Router B to realize data
communication between PC1 and PC2.
279
Figure 90 Network diagram
Configuration procedure
1.
Configure Router A:
# Set the switching mode to Re-Invite switching and the codec type to g711alaw for SIP modem
pass-through.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] default entity modem protocol pcm standard g711ulaw
# Configure VoIP voice entity 2000, and configure the IP address and the fax number of the
peer VoIP gateway as 2.2.2.2 and 2000 respectively.
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] address sip ip 2.2.2.2
[RouterA-voice-dial-entity2000] quit
# Specify 1000 as the local fax number of POTS voice entity 1000.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 1000
[RouterA-voice-dial-entity1000] line 1/0
2.
Configure Router B:
# Set the switching mode to Re-Invite switching and the codec type to g711alaw for SIP modem
pass-through.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] default entity modem protocol pcm standard g711ulaw
# Configure VoIP voice entity 1000, and configure the IP address and the fax number of the
peer VoIP gateway as 1.1.1.1 and 1000 respectively.
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] address sip ip 1.1.1.1
[RouterB-voice-dial-entity1000] quit
# Specify 2000 as the local fax number of POTS voice entity 2000.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] line 1/0
280
Configuring customizable IVR
Overview
Interactive voice response (IVR) is extensively used in voice communications. The IVR system
enables you to customize interactive operations and humanize other services. If a subscriber dials
an IVR access number, the IVR system plays the prerecorded voice prompts to direct the subscriber
about how to proceed, for example, dial a number.
Advantages
A conventional interactive voice system uses fixed audio files and operations. IVR enables you to
customize your own interactive system by adding, modifying, and removing audio files. IVR has the
following advantages.
Customizable voice prompts
Voice prompts can be saved as audio files, which must be .wav files. You can record personalized
voice prompts and upload the audio files to the voice devices. The customizable voice prompts can
be played to subscribers. The adding, modifying and removing operations in the IVR system are
simple and easy to use, and the configurations take effect instantly.
Various codecs
The IVR system supports four codecs for voice prompts: G.711alaw, G.711ulaw, G.723r5, and
G.729r8. Each kind of codec has its advantages and disadvantages:
•
G.711alaw and G.711ulaw provide high quality of voice while requiring greater memory space.
•
G.723r53 and G.729r8 provide relatively low quality of voice while requiring less memory
space.
Flexible node configuration
To simplify configuration, the IVR system uses nodes as basic units for configuration. You can define
three types of nodes: Call node, Jump node, and Service node. Each node type has a single function,
and you can combine them to realize complex functions.
•
Call node—Executes a secondary call.
•
Jump node—Jumps to another node according to the input of the subscriber.
•
Service node—Executes various operations, such as executing an immediate secondary call,
auto jumping, terminating a call, and playing an audio file.
Customizable process
The IVR system simplifies tasks such as configuring custom IVR access numbers, voice prompts,
and combinations of keys and voice prompts, making customization of interactive process easy.
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Successive jumping
The IVR process can realize successive jumping up to eight times from node to node.
Error processing methods
The IVR system provides three error processing methods: terminate the call, jump to a specified
node, and return to the previous node. To handle errors, select an error processing method for a Call
node, for a Jump node, or globally.
Timeout processing methods
The IVR system provides three timeout processing methods: terminate the call, jump to a specified
node, and return to the previous node. Select a timeout processing method for a Call node, for a
Jump node, or globally to handle the keypress timeout event.
Various types of secondary calls
The IVR system supports immediate secondary call, normal secondary call, and extension
secondary call:
•
Immediate secondary call—A subscriber makes an immediate secondary call without the
need to dial the number of the called party. Immediate secondary calls are executed by Service
nodes.
•
Normal secondary call—A subscriber makes a normal secondary call by dialing the number of
the called party. Normal secondary calls are executed by Call nodes. You can configure a node
to match the length of a number, match the terminator, or match the number.
•
Extension secondary call—A subscriber makes an extension secondary call by dialing the
extension number of the called party. Extension secondary calls are executed by Call nodes.
Hardware compatibility with customizable IVR
Customizable IVR is not available on the following routers:
•
MSR800.
•
MSR 900.
•
MSR900-E.
•
MSR 930.
•
MSR 2600.
•
MSR 30-11.
•
MSR 30-11E.
•
MSR 30-11F.
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•
MSR3600-51F.
Customizable IVR configuration task list
Task
Configuring an IVR voice
entity
Remarks
Creating an IVR voice entity
Required.
Configuring an IVR voice entity
Required.
Specifying the ID for a media resource
Required.
Configuring IVR processing methods globally
Optional.
Creating an IVR node
Configuring a Call node
Required.
Configuring a Jump node
Use one of, two of or
all of the
configurations as
needed.
Configuring a Service node
Configuring an IVR voice entity
This section describes the procedures for creating and configuring an IVR voice entity.
Creating an IVR voice entity
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program view.
dial-program
N/A
4.
Create an IVR voice entity and
enter IVR voice entity view.
entity entity-number ivr
By default, no IVR voice entity is
created.
Configuring an IVR voice entity
To configure the root node:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create an IVR voice entity
and enter IVR voice entity
view.
entity entity-number ivr
N/A
Configure the root node,
that is, the first node of the
IVR voice entity.
ivr-root node-id
By default, no root node is
configured for an IVR voice entity.
5.
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To configure the commands for an IVR voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program
view.
dial-program
N/A
4.
Create an IVR voice entity
and enter IVR voice entity
view.
entity entity-number ivr
N/A
Configure a target
match-template for the
IVR voice entity.
match-template match-string
By default, no number template is
configured for an IVR voice entity.
5.
6.
Optional.
Register numbers of the
IVR voice entity with the
H.323 gatekeeper or SIP
server.
register-number
7.
Bind the subscriber group
to the IVR voice entity.
caller-group { deny | permit }
subscriber-group-list-number
8.
Configure the calling
numbers permitted to
originate calls to the IVR
voice entity.
caller-permit calling-string
By default, the numbers of the IVR
voice entity are registered on the
H.323 gatekeeper or SIP server.
Optional.
By default, no subscriber group is
bound to an IVR voice entity, that
is, any calling number is allowed.
Optional.
By default, incoming calls are not
restricted.
Optional.
9.
Specify the codecs and
their priority levels for the
IVR voice entity.
compression { 1st-level |
2nd-level | 3rd-level | 4th-level }
{ g711alaw | g711ulaw | g723r53 |
g729r8 }
The IVR system supports four
codecs: G.711alaw, G.711ulaw,
G.723r53 and G.729r8. By default,
the codec with the first priority is
G.729r8, that with the second
priority is G.711alaw, that with the
third priority is G.711ulaw, and that
with the fourth priority is G.723r53.
The IVR voice entity does not
support the G.726 codec.
The default entity compression
command takes no effect on an
IVR voice entity.
10. Configure the description
string for the IVR voice
entity.
11. Set the DSCP value in the
ToS field in the IP packets
that carry RTP streams of
the IVR voice entity.
Optional.
description text
By default, no description is
configured for an IVR voice entity.
Optional.
dscp media dscp-value
By default, the DSCP value is ef
(101110).
Optional.
12. Configure the voice
packetization period for
different codecs.
payload-size { g711 | g723 | g729 }
time-length
The default is 20 milliseconds for a
G.711 codec, and 30 milliseconds
for G.723, and G.729 codecs.
Because the IVR voice entity does
not support g726 codecs, the
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Step
Command
Remarks
packetization periods configured
for g726 codecs on an IVR voice
entity take no effect.
The packetization periods
configured for different codecs on
an IVR voice entity take effect to
media files, and take no effect to
MOH audio input ports.
Optional.
13. Set the
maximum-call-connection
number to the IVR voice
entity.
max-call set-number
By default, no
maximum-call-connection set is
bound to an IVR voice entity (that
is, the IVR voice entity does not
belong to any
maximum-call-connection set and
there is no limitation on the number
of call connections).
Optional.
14. Configure the priority for
the IVR voice entity.
priority priority-order
15. Disable the voice entity
search function.
select-stop
16. Configure SIP
authentication
information.
user username password { cipher
| simple } password [ cnonce
cnonce | realm realm ] *
17. Change the management
state of the IVR voice
entity from up to down.
shutdown
The default is 0.
The smaller the priority value, the
higher the priority.
Optional.
Enabled by default.
Optional.
By default, no SIP authentication
information is configured in IVR
voice entity view.
Optional.
Up by default.
Specifying the ID for a media resource
A media resource can be a media resource file or a music on hold (MOH) audio input port. Enter the
corresponding media resource management view by specifying a codec. In each media resource
management view, you can specify a media resource ID for a media resource file or an MOH audio
input port.
To specify a media resource ID for a media resource:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter IVR management view.
ivr-system
N/A
4.
Enter media resource
management view.
media-file { g711alaw |
g711ulaw | g723r53 | g729r8 }
N/A
5.
Specify an ID for a media
resource (a media resource
file or an MOH audio input
set-media media-id { file
filename | moh-interface
interface-number }
By default, no ID is specified for a
media resource.
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Step
Command
Remarks
port).
Configuring IVR processing methods globally
If you do not configure any error processing method or timeout processing method for a node, it uses
the global methods to handle errors and timeout issues.
To configure IVR processing methods globally:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter IVR management view.
ivr-system
N/A
Optional.
4.
Configure IVR global
processing method for
handling subscriber input
errors.
ivr-input-error { media-play
media-id [ play-times ] | repeat
repeat-times } *
By default, the maximum number
of times permitted for input errors
is 3. The system does not play
voice prompts for input errors and
terminates the call after the
maximum number of times is
reached.
Optional.
5.
Configure IVR global
processing methods to
handle input timeout.
ivr-timeout { expires seconds |
media-play media-id
[ play-times ] | repeat
repeat-times }*
By default, the timeout time is 10
seconds, and the maximum
timeout times are 3. The system
does not play voice prompts for
the input timeout and terminates
the call after the maximum
number of times is reached.
Creating an IVR node
You can configure three types of IVR nodes: Call node, Jump node, and Service node.
Avoid the following misconfigurations:
•
No operation is configured for a node.
•
Several nodes form a loop. The subscriber has no other options except jumping around these
nodes.
•
The IVR process jumps from node to node for more than eight times.
For more information, see "Troubleshooting IVR."
To create an IVR node:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter IVR management view.
ivr-system
N/A
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Step
Command
Remarks
4.
Create an IVR voice entity node
and enter node view.
node node-id { call | jump |
service }
N/A
5.
Configure description string for
the node.
Optional.
description string
By default, no description is
configured for an IVR voice entity
node.
Configuring a Call node
Use Call nodes to configure the secondary call function. You can configure two kinds of dial plans for
a Call node: normal secondary call and extension secondary call. If you configure both dial plans for
a Call node, the extension secondary call plan takes precedence over the normal secondary call plan,
that is, after the subscriber dials the secondary call number, the system matches it with the extension
secondary call numbers first, and then matches it with the normal secondary call numbers.
To handle input errors and input timeouts, configure error processing and timeout processing
methods for a node. If you do not configure the methods, global processing methods apply.
A timeout under a Call node is different from that under a Jump node. A timeout under a Call node
can be either a timeout before the first dial or after the first dial. If the timeout happens before the first
dial, the system applies the timeout processing method. If the timeout happens after the first dial, the
system applies the input error processing method. Therefore, the timeout processing method
configured for a node is used to handle the timeout before the first dial. After the first dial, the timeout
time for the next dial is 10 seconds by default.
To configure a Call node:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter IVR management view.
ivr-system
N/A
4.
Enter Call node view.
node node-id call
N/A
•
Configure an extension
secondary call for the node
extension
extension-number call
corresponding-number
Configure the normal
secondary call number
match mode for the node
call-normal { length
number-length | matching |
terminator character }
5.
Configure a dial plan for the
call node.
•
6.
Configure a dial prefix for the
call node.
dial-prefix string
By default, no dial prefix is
configured.
7.
Specify the audio file that will
be played to the subscriber
when the node is waiting for
the subscriber to press keys.
media-play media-id
[ play-times ] [ force ]
Optional.
Configure the input error
processing method for the
input-error { end-call |
Optional.
8.
Use either method.
Optional.
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Not configured by default.
Step
node.
9.
Configure the input timeout
processing method for the
node.
Command
Remarks
goto-pre-node | goto-node
node-id } [ media-play media-id
[ play-times ] | repeat
repeat-times ] *
Not configured by default.
timeout { end-call |
goto-pre-node | goto-node
node-id } [ expires seconds |
media-play media-id
[ play-times ] | repeat
repeat-times ] *
Optional.
Not configured by default.
NOTE:
If you do not configure error processing and timeout processing methods for a node, the system
applies global processing methods. If you configure the processing methods both for the node and
globally, the configurations made for the node takes precedence.
Configuring a Jump node
You can configure the following functions for a Jump node: playing audio files, jumping to another
node, and terminating a call. To handle input errors and input timeouts, you need to configure error
processing and timeout processing methods for a Jump node. If you do not configure the methods,
the system applies global processing methods.
To configure a Jump node:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter IVR management view.
ivr-system
N/A
4.
Enter Jump node view.
node node-id jump
N/A
5.
Configure the node to
execute the jump operation
based on the input of the
subscriber.
user-input character { end-call |
goto-node node-id |
goto-pre-node }
Not configured by default.
Specify the audio file that will
be played to the subscriber
when the node is waiting for
the subscriber to press keys.
media-play media-id
[ play-times ] [ force ]
Configure the input error
processing method for the
node.
input-error { end-call |
goto-pre-node | goto-node
node-id } [ media-play media-id
[ play-times ] | repeat
repeat-times ] *
Configure the input timeout
processing method for the
node.
timeout { end-call |
goto-pre-node | goto-node
node-id } [ expires seconds |
media-play media-id
[ play-times ] | repeat
repeat-times ] *
6.
7.
8.
288
Optional.
Not configured by default.
Optional.
Not configured by default.
Optional.
Not configured by default.
NOTE:
You can configure input timeout processing and input timeout processing methods as needed. If you
do not configure those methods for a node, the system applies global processing methods. If you
configure the processing methods both for the node and globally, the configurations made for the
node takes precedence.
Configuring a Service node
You can configure the following functions for a Service node: playing audio files, jumping to another
node, executing immediate secondary call, and terminating a call.
You can configure at most three functions for a Service node and use the select-rule
operation-order command to specify the execution order of the functions. For example, if you
specify the order by using the select-rule operation-order 2 1 3 command, the execution order of
the functions is 2->1->3. If you do not specify the execution order, the default execution order will be
applied, that is: 1->2->3.
If an executed function is to jump to another node or to terminate a call, the rest one or two functions
are not executed. For more information, see "Service node configuration example 2."
Because a Service node has no need to wait for subscriber input, you do not need to configure the
error processing and timeout processing methods for a Service node.
To configure a Service node:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter IVR management view.
ivr-system
N/A
4.
Enter Service node view.
node node-id service
N/A
5.
Specify the execution order
of the configured functions.
select-rule operation-order
1st-operation 2nd-operation
3rd-operation
Optional.
Configure functions for the
Service node.
operation number
{ call-immediate call-number |
end-call | goto-node node-id |
goto-pre-node | media-play
media-id [ play-times ] }
6.
The default execution order is
1->2->3.
Optional.
Not configured by default.
Displaying and maintaining customizable IVR
Task
Command
Remarks
Display the IVR playing
info