SIP - Michael McNamara

AT&T VoEVPN Service
Nortel CS 1000 (Release 5.50J) SIP Configuration Guide
AT&T VOIP
Nortel CS 1000 (Release 5.00W / 5.50J) SIP
Configuration Guide
For Use with AT&T
VoEVPN Services
NN10000-108
Issue 2.1
02/04/2010
CS 1000
*Nortel, Nortel Networks, Nortel (Logo), the Globemark, CS1000 are trademarks of Nortel Networks.
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AT&T VoEVPN Service
Nortel CS 1000 (Release 5.50J) SIP Configuration Guide
TABLE OF CONTENTS
1
2
3
4
5
6
7
Introduction................................................................................................................. 4
1.1
Pre-IP PBX Configuration Activity .................................................................... 4
1.2
Customer Questions ............................................................................................ 4
1.3
Trouble Reporting............................................................................................... 4
1.4
Document Feedback............................................................................................ 5
1.5
Document Change History.................................................................................. 5
Special Notes .............................................................................................................. 6
Overview..................................................................................................................... 7
Configuration Guide ................................................................................................. 10
4.1
Nortel CS 1000 Version.................................................................................... 10
4.1.1
Nortel CS 1000 Release 5.5 ...................................................................... 10
4.2
Nortel CS 1000 Patches .................................................................................... 11
4.2.1
Patches for Nortel CS 1000 Release 5.5 ................................................... 11
4.3
Node Configuration .......................................................................................... 13
4.3.1
IP Codecs .................................................................................................. 13
4.3.2
LAN Configuration................................................................................... 14
4.3.3
SIP Gateway Settings................................................................................ 15
4.3.4
SIP URI Map............................................................................................. 15
4.3.5
Signaling Server Properties....................................................................... 16
4.4
VoIP Trunking .................................................................................................. 18
4.4.1
Creating D-Channels................................................................................. 18
4.4.2
Incoming Digit Conversion and CLID Trees............................................ 20
4.4.2.1 Creating IDC Trees ............................................................................... 20
4.4.2.2 Creating CLID Trees............................................................................. 21
4.4.3
Creating Routes......................................................................................... 22
4.4.4
Adding Trunks to the Specific Route........................................................ 24
4.4.4.1 Disabling Media Security on Virtual Trunks........................................ 25
4.4.5
Creating a Digit Manipulation Index (DMI)............................................. 26
4.4.6
Creating Route List Block Indices (RLI).................................................. 26
4.4.7
Allowing/Restricting Numbering Plan Area (NPA) Codes ...................... 27
4.4.8
Configuring Special Numbers................................................................... 28
Basic Monitoring and Call Tracing........................................................................... 29
5.1
Viewing Registered Sets on CS 1000 ............................................................... 29
5.2
Active Call Information .................................................................................... 29
5.3
SIP Call Tracing................................................................................................ 30
Table of Acronyms ................................................................................................... 32
Appendix A: Downloading CS 1000 Patches........................................................... 34
*Nortel, Nortel Networks, Nortel (Logo), the Globemark, CS1000 are trademarks of Nortel Networks.
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TABLE OF FIGURES
Figure 1: AT&T VoEVPN Architecture............................................................................. 7
Figure 2: AT&T VoEVPN and private networking............................................................ 8
Figure 3: CS 1000 Release 5.5 Call Server software........................................................ 10
Figure 4: CS 1000 Release 5.5 Signaling Server software ............................................... 10
Figure 5: Viewing CS 1000 Signaling Server patches...................................................... 11
Figure 6: Viewing CS 1000 Call Server patches .............................................................. 12
Figure 7: Node configuration screen................................................................................. 13
Figure 8: IP codecs............................................................................................................ 14
Figure 9: RTP/RTCP port for IP sets................................................................................ 15
Figure 10: SIP GW settings .............................................................................................. 15
Figure 11: SIP URI map ................................................................................................... 16
Figure 12: Signaling Server properties ............................................................................. 17
Figure 13: D-channel basic configuration......................................................................... 18
Figure 14: D-channel basic options .................................................................................. 19
Figure 15: D-channel advanced options ........................................................................... 19
Figure 16: Incoming digit conversion tree........................................................................ 20
Figure 17: Converting incoming 10 digits to DN ............................................................. 21
Figure 18: Creating CLID table for DODs matching DN................................................. 21
Figure 19: Creating CLID table for DODs not matching DN .......................................... 22
Figure 20: Route configuration......................................................................................... 23
Figure 21: IDC for route ................................................................................................... 24
Figure 22: Adding multiple trunks.................................................................................... 24
Figure 23: Trunk basic configuration ............................................................................... 25
Figure 24: Trunk advanced configuration......................................................................... 25
Figure 25: Disabling media security on the trunk............................................................. 25
Figure 26: Digit Manipulation Index ................................................................................ 26
Figure 27: Route list block parameters ............................................................................. 26
Figure 28: Route list block, data entry index.................................................................... 27
Figure 29: NPA codes....................................................................................................... 27
Figure 30: Special numbers .............................................................................................. 28
Figure 31: Displaying registered sets to a specific node................................................... 29
Figure 32: Displaying registered sets to a specific Signaling Server................................ 29
Figure 33: Displaying registered sets based on phone models ......................................... 29
Figure 34: Active call information.................................................................................... 30
Figure 35: Sample SIPCallTrace output ........................................................................... 31
*Nortel, Nortel Networks, Nortel (Logo), the Globemark, CS1000 are trademarks of Nortel Networks.
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AT&T VoEVPN Service
Nortel CS 1000 (Release 5.50J) SIP Configuration Guide
1 Introduction
This document provides a configuration guide to assist Nortel Networks CS 1000
administrators in connecting to AT&T Voice over Enhanced Virtual Private Network
(VoEVPN) service.
This configuration guide pertains specifically to the AT&T Voice over EVPN service
and is not intended for configuring a CS 1000 system for new installation.
1.1 Pre-IP PBX Configuration Activity
This guide assumes that the administrator is knowledgeable in CS 1000 IP PBX
programming and operations.
An important tool that the administrators should have at their disposal prior to testing
their IP PBX with Voice over EVPN service is a network protocol analyzer. Such
software can be used to run traces on problem calls so the information can be shared with
equipment and network engineers. There is a free version of such software that can be
obtained at http://www.wireshark.org/.
A second alternative that customers may use is TCPDUMP which can be found on most
UNIX and Linux systems. To use this software the customer should have Wireshark or
TCPDUMP loaded on a server that is connected to a LAN switch or hub that can monitor
both the signaling and media packets on any calls between the customer PBX and the
Voice over EVPN managed router. Please note, however, that AT&T does not offer,
warrant, or support this software, and any use of the Wireshark or TCPDUMP software is
entirely at the customer’s own risk.
1.2 Customer Questions
Section 4 of this guide provides screen shots and instructions for the configuration of
your IP PBX. Should you have questions regarding these instructions, please call Brian
Stegemoller at +1 (972) 685-6629; ((972) 745-5139 after 2/22/2010). When calling this
number please have the following information available:
ƒ
ƒ
ƒ
ƒ
ƒ
Company name
Company location
Administrator name and phone number
IP PBX name and software version
Customer Configuration Guide – Issue number and date
1.3 Trouble Reporting
Nortel and AT&T will make every effort to quickly resolve reported troubles. The time
required for trouble shooting can be reduced if the customer has the necessary detailed
*Nortel, Nortel Networks, Nortel (Logo), the Globemark, CS1000 are trademarks of Nortel Networks.
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information available when reporting a problem. Prior to reporting a problem please
provide a Wireshark or TCPDUMP trace of the failed call.
1.4 Document Feedback
IP PBX administrators who would like to provide feedback on the contents of this
document should send it to Brian Stegemoller at (brianstegemo@avaya.com) with a copy
to Al Chee (alchee@avaya.com) and Steven Chen (stevenchen@avaya.com).
1.5 Document Change History
Draft 0.1
Draft 0.2
Issue 1.0
Issue 2.0
Issue 2.1
12/1/2008; Initial draft
12/11/2008; ensured that VoEVPN
acronym is used throughout document.;
added notation for failed CallPilot voice
mail retrieval and added emergency 911
limitation in Section 2; inserted Universal
Trunk Card in equipment list and changed
G.729 codec supported, from G.729AB to
G.729 30 ms payload size in Section 3.
12/17/2008; General availability for issue
1.0 of CCG guide.
02/04/2009; Corrected Media Gateway
Controller (MGC) Codec from G.729B to
G.729A and VAD value unchecked.
02/04/2010
Updated Contact Information to Reflect
AVAYA Merger
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2 Special Notes
VoEVPN Managed Access Router Information
All the VoEVPN access router fixes associated with the CS1000 are included in Cisco
release IOS 12.4.(15)T.6 and later.
Voice Mail Retrieval from IOS Cisco Gateway to Nortel CallPilot Fails.
When compressed RTP is enabled on access routers, on-net voice mail retrieval to the
Nortel CallPilot does not work properly. It is recommended to deploy larger bandwidth
circuits if remote CallPilot voice mail retrieval is desired. Alternatively, the PSTN can
also be used to access CallPilot voice mail systems.
Emergency 911/E911 Service Limitations
All emergency 911 calls should be routed through PSTN trunks at each CS1000 IP PBX
locations.
.
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Nortel CS 1000 (Release 5.50J) SIP Configuration Guide
3 Overview
This section provides a service overview of the Nortel Networks Communication Server
1000 Release 5.5 (CS1000 R5.5) IP PBX integration with AT&T Voice over Enhanced
Virtual Private Network (VoEVPN) service. This will enable the CS1000 to place VoIP
Gateway to Gateway, on-net and off-net PSTN (hop-off only) calls using the SIP
(Session Initiation Protocol) protocol.
PSTN
PSTN
AT&T
CCE
AT&T
NGBE
Public Side
Private Side
Switch
NAT
Firewall
Nortel CS 1000
IP PBX
AT&T
Managed
router
Cisco router
With
Voice GW
Legacy Circuit
PBX
Public Side
AT&T Common
IP Backbone
AT&T
Managed
router
PSTN
PSTN
Public Side
NAT
Firewall
Switch
Nortel CS 1000
IP PBX
Figure 1: AT&T VoEVPN Architecture
The Nortel CS 1000 customer premises site shall consist of the following components:
ƒ
ƒ
Nortel IP Phone 2000 series, 1100 series, IP Softphone 2050 (Release
3.00.0197 and up)* – These phones use the Nortel UNIStim signaling
protocol to communicate to the Nortel CS 1000 IP PBX for call feature and
routing support. These phones can be connected to a Nortel Ethernet switch
(ES 470, ERS 5520, etc.) that supplies in-line power (IEEE 802.3af) to the
phones.
Nortel CS 1000 IP PBX – This unit consists of the following:
o Dedicated Signaling Server (can be COTS, ISP 1100, or CP-PM)
o Call Server (can be CP-PIV or CP-PM cards)
o Media Gateway Controller (MGC) card to provide Digital
Signaling Processor (DSP) resources for connecting IP and Time
Division Multiplexing (TDM) devices together and for advanced
applications such as conferences and voicemail access
o CallPilot voicemail system (optional)
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o Digital Line Card (DLC) for Meridian digital sets (optional)
o Analog Message Waiting Line Card (AM/WLC) for connection to
fax machines and analog sets (optional)
o 8- Port Universal Trunk Card (Analog trunks to PSTN)
o TMDI card as a PRI/T1 trunk to the PSTN (optional)
IMPORTANT
A dedicated Signaling Server will be needed for connectivity to
AT&T VoEVPN service. Private MCDN features would require an
additional Signaling Server. See figure below:
AT&T
VoEVPN
Signaling
Servers
CS1000
Private
Network
CS1000
PSTN
CS1000
PSTN
Site 3
Site 1
PSTN
Site 2
Figure 2: AT&T VoEVPN and private networking
Please note that this guide does not describe the procedures to configure private
MCDN functionality with the CS 1000. This guide only pertains to the Signaling
Server connected to AT&T Voice over EVPN.
The following routing scenarios are supported by the Nortel CS 1000 IP PBX and DO
NOT use the AT&T Call Control:
ƒ
ƒ
ƒ
ƒ
Local Nortel CS 1000 phone to local Nortel CS 1000 phone
Inbound PSTN to Nortel Networks CS1000 phone
Outbound local PSTN calls from the Nortel Networks CS1000 phones.
Outbound local N11 (i.e. 411, 911) calls from the Nortel Networks CS1000
phones
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The following routing scenarios are supported by the CS 1000 IP PBX and DO use the
AT&T Call Control. For voice calls, the G.729A codec and 30 ms payload shall be used:
ƒ
ƒ
ƒ
ƒ
Nortel CS 1000 phones to PSTN (domestic US and international)
Nortel CS 1000 phones to legacy PBX site with Cisco gateway
Legacy PBX site with Cisco gateway to Nortel CS 1000 phones
Nortel CS 1000 phones at one Nortel CS 1000 IP PBX site to Nortel CS 1000
phones at another Nortel CS 1000 IP PBX site.
Fax was tested and is supported on the CS 1000 using the T.38 fax protocol through the
AT&T VoEVPN network to/from (except PSTN) the following:
ƒ
ƒ
ƒ
PSTN (Outbound only)
Legacy PBX site with Cisco gateway
Another CS 1000 IP PBX site
* The new release of the Nortel IP Softphone 2050 (Release 3.00.0197) is supported with
CS 1000 and AT&T VoEVPN.
*Nortel, Nortel Networks, Nortel (Logo), the Globemark, CS1000 are trademarks of Nortel Networks.
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Nortel CS 1000 (Release 5.50J) SIP Configuration Guide
4 Configuration Guide
This configuration guide specifies the Nortel CS 1000 screens that must be configured
and updated to support the AT&T VoEVPN service.
4.1 Nortel CS 1000 Version
The Nortel CS 1000 IP PBX software version 5.5 is required.
4.1.1 Nortel CS 1000 Release 5.5
The Nortel Networks CS 1000 Call Server must be running release 5.50J. You can check
the version of CS 1000 by viewing the following screen on the Home page:
Figure 3: CS 1000 Release 5.5 Call Server software
The CS 1000 Signaling Server must be running release sse-5.50.12:
Figure 4: CS 1000 Release 5.5 Signaling Server software
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4.2 Nortel CS 1000 Patches
To verify installed/applied patches on the Signaling Server, go to
System » Software » Servers and Media Cards, select the appropriate Element
(Signaling Server) and Platform types, and click on the “PSTAT” button.
To download CS 1000 patches, please refer to Section 7, Appendix A: Downloading
CS 1000 Patches.
Figure 5: Viewing CS 1000 Signaling Server patches
4.2.1 Patches for Nortel CS 1000 Release 5.5
The following patches are needed for the Signaling Server:
Patch ID
MPLR22452
MPLR22968
MPLR24785
MPLR25982
MPLR23632
Description
Don't attach MCDN message for outgoing INVITE
TO/FROM URI Incorrect - Replace domain population in the
FROM field to the Node IP Address of the Signaling Server.
Updated version 2 corrected PRACK issue.
If the privacy value is set to none, then privacy header is removed
** IMPORTANT: Ensure that calling line privacy feature is
disabled on the CS 1000 system as well as on the individual sets. **
No speech path if 183 session in progress comes before 180 Ringing
(need this for outbound calls to call prompters and AT&T wireless
cell phones)
Null values should be allowed for Public E.164/National or
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MPLR27236
Subscriber fields
New functionality for handling “maxptime” for outgoing calls. The
CS1000 will now use the configured codec payload value e.g. 20ms,
30ms, etc.
To verify installed/applied patches on the Call Server, go to System » Software » Call
Server, select “PEP Status (PSTAT)” for the command, and click “Submit.”
Figure 6: Viewing CS 1000 Call Server patches
Currently no patches are required for the Call Server:
Patch ID
None
NOTE
Description
In order to view and install CS 1000 Call Server patches, a level 2
user with PDT2 privileges must be logged in.
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4.3 Node Configuration
Add or edit a node in the System » IP Network » Node Configuration menu with the
following configuration as noted in the below screenshots:
Figure 7: Node configuration screen
4.3.1
IP Codecs
The following codecs should be enabled:
ƒ G711 with Voice payload size to 30 ms/frame
ƒ G729A with Voice payload size to 30 ms/frame, VAD is unchecked
ƒ T38 FAX
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Figure 8: IP codecs
SIP (VTRK) trunks should be in a ZONE that uses “Best Bandwidth (BB).” This will
ensure that G.729 codec is offered first.
Ensure that the IP codec settings in this section are also configured on the Media
Gateway Controller (MGC) card under the VGW and IP phone codec profile section in
the System » IP Network » Media Gateways » IPMG x x Property
Configuration » IPMG x x Media Gateway Controller (MGC) Configuration menu.
For interoperability with AT&T Voice over EVPN, use G.729A (under Codec G729A
sub-section, ensure that VAD is unchecked).
4.3.2
LAN Configuration
All IP sets use the same port ranges for media. This is specified in the LAN
configuration section, under the Telephony LAN (TLAN) configuration sub-section. In
the figure below, the starting port 29100 was entered. Note: for VoEVPN access routers
requiring compression, select a starting port from 16384 through 32768.
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Figure 9: RTP/RTCP port for IP sets
4.3.3
SIP Gateway Settings
AT&T VoEVPN service does not support TLS Security, thus the Security Policy should
be set to “Security Disabled.” Enter the Signaling Server’s TLAN IP address for the
Primary Proxy or Redirect (TLAN) IP address, ensure the Port used is 5060, and the
Transport Protocol used is UDP.
Figure 10: SIP GW settings
4.3.4
SIP URI Map
Ensure that all parameters for the SIP URI Map are left blank; otherwise, calls to
AT&T VoEVPN service will fail. This can be done with MPLR23632 loaded.
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Figure 11: SIP URI map
4.3.5
Signaling Server Properties
For the Signaling Server that will be used to peer with the AT&T VoEVPN network,
configure the Virtual Trunk TPS to “SIP only”, ensure that the SIP Proxy/Redirect
Server option is checked, Local SIP TCP/UDP Port to Listen to is 5060, SIP Domain
name is the IP address of the AT&T SIP Proxy Server, and Enable Gatekeeper option is
unchecked.
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Figure 12: Signaling Server properties
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4.4 VoIP Trunking
Voice over IP (VoIP) lines, are signaling channels that simulate how CO lines work.
However, VoIP lines transmit data to the IP network over a LAN or IP network rather
than over physical lines.
To create VoIP trunks on the CS 1000 to the AT&T VoEVPN service, the following
steps need to be executed:
ƒ Creating D-Channels
ƒ Creating Incoming Digit Conversion (IDC) and Calling Line Identification
(CLID) trees
ƒ Creating Routes
ƒ Adding Trunks to the specific Route
ƒ Creating a Digit Manipulation Index (DMI)
ƒ Creating Route List Block Indices (RLI)
ƒ Allowing/Restricting NPA Codes
ƒ Configuring Special Numbers
4.4.1
Creating D-Channels
Call signaling on the CS 1000 resides on the D-channels. A route will be mapped to this
D-channel, in which the Signaling Server will send call signaling to the AT&T VoEVPN
service using the specified D-channel. Under the Routes and Trunks » D-Channels
menu, add a D-channel with the following configuration:
Figure 13: D-channel basic configuration
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Figure 14: D-channel basic options
Figure 15: D-channel advanced options
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4.4.2
Incoming Digit Conversion and CLID Trees
Recommended best practices for planning DNs with DIDs is to map the last four digits of
the Direct Inward Dialing (DID) number to the CS 1000 DNs. However, there will be
scenarios where DNs may not match with the AT&T-provided DID number extensions.
In this case, the Incoming Digit Conversion (IDC) and Calling Line Identification (CLID)
trees will be used.
4.4.2.1 Creating IDC Trees
The IDC tree will allow incoming digits to be converted to specified local extensions.
This is done in Dialing and Numbering Plans » Incoming Digit
Translation » Customer 00 » Digit Conversion Tree x Configuration menu, clicking
“Add,” then entering incoming and converted digits; see figure below:
Figure 16: Incoming digit conversion tree
NOTE
Up to 8 digits can be entered in for the Incoming Digits field in
Element Manager. To enter more, read below.
If AT&T VoEVPN service sends 732-216-2779 to the CS 1000, the CS 1000 will use
entry 2 to strip 732-216-2, convert the digits to 2, append the remainder of the digits and
ring the set with DN 2779.
In the case where the full 10 digit number needs to be converted to a DN, the
administrator will have to use the Call Server CLI. For example, to convert 732-3680430 to 2001, go to ld 49:
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>ld 49
DGT000
MEM AVAIL: (U/P): 99201863
USED U P: 5027918 40040
TOT: 104269821
DISK SPACE NEEDED: 41 KBYTES
REQ chg
TYPE idc
CUST 0
<customer number>
DCNO 1
<digit conversion tree number>
IDGT 7323680430
<incoming digit>
7323680432 2001
IDGT
<converted digit>
Figure 17: Converting incoming 10 digits to DN
4.4.2.2 Creating CLID Trees
The same concept is applied to Direct Outward Dialing (DOD) numbers for outbound
calling numbers. This is done in the Call Server CLI, ld 15. For creating CLID tables
where the extensions of the phones match the DID/DOD, see Section 4.4.2.2.1; for nonmatching extensions, see Section 4.4.2.2.2.
4.4.2.2.1 Matching Extensions
For the example of using DN 2779 to 732-216-2779 (last four digits of extension match
DN):
>ld 15
MEM AVAIL: (U/P): 99201833
USED U P: 5027918 40070
TOT: 104269821
DISK SPACE NEEDED: 41 KBYTES
REQ: chg
TYPE: net
CUST 0
OPT
AC2
FNP
CLID yes
SIZE
INTL
ENTRY <enter #>
HNTN 732
ESA_HLCL
ESA_INHN
ESA_APDN
HLCL 216
DIDN YES
* use when DN matches DOD extension
HLOC
LSC
CLASS_FMT DN
ENTRY # SAVED!
Figure 18: Creating CLID table for DODs matching DN
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Be sure when you configure the keys for the IP set in ld 11 with the above configuration,
use the appropriate CLID table entry #. For example, KEY 0 scr 2779 # will allow the
CS 1000 to send calling party number of 732-216-2779 out to AT&T IP Flexible Reach
service.
4.4.2.2.2 Non-matching Extensions
For the example where DN 2001 needs to be converted to DOD 732-368-0430:
>ld 15
MEM AVAIL: (U/P): 99201833
USED U P: 5027918 40070
TOT: 104269821
DISK SPACE NEEDED: 41 KBYTES
REQ: chg
TYPE: net
CUST 0
OPT
AC2
FNP
CLID yes
SIZE
INTL
ENTRY <enter #>
HNTN 732
ESA_HLCL
ESA_INHN
ESA_APDN
HLCL 3680430
DIDN NO
* use when DN DOES NOT match DOD extension
HLOC
LSC
CLASS_FMT DN
ENTRY # SAVED!
Figure 19: Creating CLID table for DODs not matching DN
Be sure when you configure the keys for the IP set in ld 11 with the above configuration,
use the appropriate CLID table entry #.
4.4.3
Creating Routes
Once the D-Channel is created, routes will be created to map to the D-Channel and the
IDC trees, if applicable. The route will be configured with SIP trunking to the AT&T IP
Flexible Reach service. At the Routes and Trunks » Routes and Trunks screen, add a
route:
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Figure 20: Route configuration
Ensure of the following settings:
ƒ Trunk Type is TIE
ƒ Incoming and Outgoing trunk is Incoming and Outgoing (IAO)
ƒ The route is a virtual trunk route is checked
ƒ Node ID of signaling server is set to Signaling Server peering with AT&T
Voice Over EVPN service
ƒ Zone for codec selection and bandwidth management is set to a zone for
the virtual trunks
ƒ Protocol ID for the route is SIP
ƒ ISDN option is checked
ƒ Mode of operation is ISLD
ƒ D channel number is D-channel created in Section 4.4.1
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Furthermore, under the same screen, in Basic Route Options, ensure that IDC is checked
and enter in IDC tree numbers for both Day and Night IDC trees.
Figure 21: IDC for route
4.4.4
Adding Trunks to the Specific Route
Trunks can now be added to associate with a given route. Back to the Routes and
Trunks » Routes and Trunks screen, for the route created in Section 4.4.3, click “Add
trunk.”
NOTE:
To add multiple trunks, select the number of trunks to add from the dropdown menu in MTINPUT.
Figure 22: Adding multiple trunks
Under the Basic Configuration, ensure the following:
ƒ TYPE is IPTI
ƒ Enter TN for the trunks
ƒ XTRK is VTRK
ƒ RTMB in following notation <route number in 4.4.3> <number>
ƒ CDEN is 8D
ƒ STRI, STRO is IMM
ƒ CHID is a different number for each trunk
ƒ INC set to YES
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Figure 23: Trunk basic configuration
Under the Advanced Trunk Configurations, ensure that SUPN and STYP are checked
and PIP, respectively:
Figure 24: Trunk advanced configuration
4.4.4.1 Disabling Media Security on Virtual Trunks
In conference and transfer call scenarios where the CS 1000 sends a SIP re-INVITE to an
endpoint via the virtual trunks to AT&T, if “Media Security” is enabled on these trunks,
calls will fail. In order to fully interoperate with AT&T IP Flexible Reach service,
“Media Security” must be set to “Media Security Never (MSNV)” under the Routes and
Trunks » Routes and Trunks » Customer 0, Route x, Trunk x Property
Configuration » Class of Service Configuration for each virtual trunk peering with
AT&T IP Flexible Reach service.
Figure 25: Disabling media security on the trunk
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4.4.5
Creating a Digit Manipulation Index (DMI)
In order for outbound calls (more specifically, special numbers) to send the correct
information to the AT&T network, ensure that a Digit Manipulation Index (DMI) is
created in Dialing and Numbering Plans » Electronic Switched Network (ESN) »
Customer 00 » Network Control & Services » Digit Manipulation Block List » Digit
Manipulation Block with the following settings (in this example, DMI 15 is used):
Figure 26: Digit Manipulation Index
This will ensure that the proper calling line ID is sent to the AT&T network, and thus
allowing calls to special numbers to complete.
4.4.6
Creating Route List Block Indices (RLI)
When an outbound call is made, based on the number dialed, a route list block index is
used to determine the route and D-channel used for call signaling.
To create a RLI, go to Dialing and Numbering Plans » Electronic Switched Network
(ESN) » Customer 00 » Network Control & Services » Route List Blocks, and add a
route list index.
Figure 27: Route list block parameters
Add a “Data Entry Index” and ensure that ROUT is set to the route number created in
4.4.3 and DMI is set to the DMI created in 4.4.5:
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Figure 28: Route list block, data entry index
4.4.7
Allowing/Restricting Numbering Plan Area (NPA) Codes
Add the allowed NPA codes for the CS 1000 in the Dialing and Numbering
Plans » Electronic Switched Network (ESN) » Customer 00 » Numbering Plan (NET)
> Access Code 1 » Numbering Plan Area Code List menu, and map the Route List
Index to the RLI configured in the previous section. Below is an example for area code
732:
Figure 29: NPA codes
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4.4.8
Configuring Special Numbers
Special numbers are numbers that do not follow the NPA dial plans. Examples are 1-800,
N11 (411, 911, etc.), and international calls. To configure these, go to Dialing and
Numbering Plans » Electronic Switched Network (ESN) » Customer
00 » Numbering Plan (NET) > Access Code 1 » Special Number List, and add
allowed special numbers. Map these numbers to the RLI in Section 4.4.6; below is an
example for 411:
Figure 30: Special numbers
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5 Basic Monitoring and Call Tracing
The following procedures below can be used to monitor and trace calls on the CS 1000.
The CS 1000 has an extensive suite of diagnostic procedures, and is out of the scope of
this document. For more information on advanced diagnostics, please refer to the Nortel
CS 1000 Release 5 Technical Documentation.
5.1 Viewing Registered Sets on CS 1000
The following Call Server CLI command in ld 96 can be used to display registered sets:
=> ecnt node
Node: 1001
Number of Registered Ethersets
: 5
Figure 31: Displaying registered sets to a specific node
=> ecnt ss
Signaling Server: SS_1001 IP: 192.12.0.10
Number of Registered Ethersets
: 5
Figure 32: Displaying registered sets to a specific Signaling Server
=> ecnt modl
2004P2: IP Phone 2004 Phase 2
Number of IP phones: 2
1140E: IP Phone 1140E
Number of IP phones: 3
Figure 33: Displaying registered sets based on phone models
5.2 Active Call Information
Administrators can view active call information for the specific DN while on a call. Go
to ld 80 in the Call Server CLI and enter the following command: trac <customer #>
<DN>. Information such as IP addresses, codecs, calling/called party numbers, etc. is
displayed:
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.trac 0 2001
ACTIVE
VTN 096 0 01 01
ORIG
VTN 096 1 02 00
VTRK IPTI RMBR 16 1 INCOMING VOIP GW CALL
FAR-END SIP SIGNALLING IP: 135.25.29.135
FAR-END MEDIA ENDPOINT IP: 135.25.29.70 PORT: 16390
FAR-END VendorID: Cisco-SIPGateway/IOS-12.x
TERM
VTN 096 0 01 01 KEY 0 SCR MARP CUST 0 DN 2001 TYPE 1140
MEDIA ENDPOINT IP: 172.16.6.101 PORT: 28802
MEDIA PROFILE: CODEC G.729A NO-LAW PAYLOAD 20 ms VAD OFF
RFC2833: RXPT 96
TXPT 96
DIAL DN 2001
MAIN_PM ESTD
TALKSLOT ORIG 66
TERM 2
EES_DATA:
NONE
QUEU NONE
CALL ID 0 18822
---- ISDN
CALL REF #
BEARER CAP
HLC =
CALL STATE
CALLING NO
CALLED NO
ISL CALL (ORIG) ---= 400
= VOICE
=
=
=
10
ACTIVE
17323681000 NUM_PLAN:E164
7323680430 NUM_PLAN:UNKNOWN
TON:INTERNATIONAL ESN:UNKNOWN
TON:UNKNOWN
ESN:UNKNOWN
Figure 34: Active call information
5.3 SIP Call Tracing
When a call is being placed to/from the CS 1000, the SIP messages can be outputted to
the screen when logged into the Signaling Server CLI. The SIPCallTrace command can
be turned on to view the SIP messages. The SIPTraceLevel command can be set to
either 0 or 1; 0 for less-detailed output, 1 for more details.
NOTE
Nortel strongly recommends that this command be left off during
normal operations; the command to disable is as follows:
SIPCallTrace off.
Below is a sample output:
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oam> SIPTraceLevel 1
oam> SIPCallTrace on
oam>
oam> 02/10/2007 16:48:13 LOG0003 SIPNPM: sipNpmPrivacyHdrBuild: : PATCH Privac
y Value was equal to NONE, so header not added PATCH
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace: This is Outgoing Message
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
Message: Outgoing method INVITE(0) chid: 27 Called num: 17324208823 Far End
Sig
naling IP: 172.16.6.111:5060 Transport:UDP CSeq: 1 INVITE
From: "Al Chee"<sip:7323680430@172.16.6.110;user=phone>
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:17324208823@135.25.29.135;user=phone>
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
Media Info: 172.16.6.101 Codecs: G729(18) G711 U-Law(0) G711 A-Law(8)
Dynamic(9
6) Dynamic(111) Payload: 20 ms Media State: SIPNPM_MEDIA_SENDRECV
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace: This is Incoming Message
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
Message: Incoming response 302 Moved Temporarily chid: 27 Called num:
1732420882
3 Far End Signaling IP: 135.25.29.135:65535 Transport:UDP CSeq: 1 INVITE
From: "Al Chee"<sip:7323680430@172.16.6.110;user=phone>
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:17324208823@135.25.29.135;user=phone>
Contact: <sip:17324208823@135.25.29.135;user=phone>
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace: This is Outgoing Message
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
Message: Outgoing method ACK(1) chid: 27 Called num: 17324208823 Far End
Signal
ing IP: 172.16.6.111:5060 Transport:UDP CSeq: 1 ACK
From: "Al Chee"<sip:7323680430@172.16.6.110;user=phone>
02/10/2007 16:48:13 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:17324208823@135.25.29.135;user=phone>
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31
Figure 35: Sample SIPCallTrace output
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6 Table of Acronyms
Acronym
AM/WLC
COTS
Definition
Analog Message Waiting Line Card: provides talk battery
and signaling for regular 2-wire common battery 500-type
(rotary dial) and 2500-type (Digitone dial) telephones and
key telephone equipment. This card also supports
message waiting indication for sets equipped with the
message waiting feature.
Commercial Off-The-Shelf: standard server hardware
manufactured by a third-party; the CS 1000 supports the
following COTS servers – IBM xSeries 306m (types
8848, 8491) and HP ProLiant DL320 G4 servers.
CP-PIV
Common Process Pentium IV: processor for the
Call/Signaling Server on a legacy large system (CS
1000M), is also supported on CS 1000E systems.
CP-PM
Common Processor Pentium Mobile: the main processor
for the Call Server, controlling all call processing and
telephony services. It also provides the system memory
required to store operating software and customer data.
This is the default processor for the CS 1000E.
Note: the Signaling Server can also be a CP-PM card.
CS 1000
DLC
Communications Server 1000
Digital Line Card: provides a multiplexed voice, data, and
signaling path to and from a digital terminal (telephone)
over a 2-wire full duplex 512 kHz Time Compression
Multiplexed (TCM) digital link.
ISP 1100
Legacy Nortel Signaling Server running VxWorks realtime operating system. In order for compatibility with
Release 5.0, the ISP 1100 must contain at least 1 GB of
memory.
MCDN
Meridian Customer Defined Network: private voice
networking functionalities/features in a Nortel IP PBX
environment
MGC
Media Gateway Controller: provides Digital Signaling
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Processor (DSP) resources for connecting IP and Time
Division Multiplexing (TDM) devices together and for
advanced applications such as conferences and voicemail
access
TLAN
Telephony Local Area Network: network interfacing with
AT&T IP Flexible Reach service
TMDI
PRI circuit card to interface with PSTN (if needed)
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7 Appendix A: Downloading CS 1000 Patches
Go to the Nortel website at http://www.nortel.com. Under “Support & Training,” select
“Software Downloads.”
This will take you to the Nortel Technical Support page. Under “Documentation,
Software, and Bulletins,” select “Voice, Multimedia & Unified Communications.”
Under “Communication & Application Services,” go to the “Communication Server
1000E” link, which will load the product page. Under the “Software” section, select the
“Patches” link.
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To search for a specific patch, enter the patch number (i.e. 23267 was entered), and select
the “Number” option.
When the search is complete, select the appropriate patch by clicking on the link, which
will load the download page.
BE SURE TO DOWNLOAD THE CORRECT PATCH FOR THE APPROPRIATE
SOFTWARE LOAD AND HARDWARE TYPE.
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This Customer Configuration Guide ("CCG") is offered as a convenience to Nortel
Networks’ (“Nortel”) customers. The specifications and information regarding the
product in this CCG are subject to change without notice. All statements, information,
and recommendations in this CCG are believed to be accurate but are presented without
warranty of any kind, express or implied, and are provided “AS IS”. Users must take full
responsibility for the application of the specifications and information in this CCG.
IN NO EVENT SHALL NORTEL OR ITS AGENTS OR SUPPLIERS BE LIABLE
FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, PUNITIVE, OR INCIDENTAL
DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR
DAMAGE ARISING OUT OF THE USE OR INABILITY TO USE THIS CCG, EVEN
IF NORTEL OR ITS AGENTS, OR SUPPLIERS HAVE BEEN ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
- End of Document -
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