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© 2008 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com
Page 1 of 45 EDCS-966105 Rev# 1
Application Note
Avaya S8500 Release CM 5.2.1 using SIP trunk to Cisco Unified
Communications Manager Release 8.5
December 20, 2010 – Rev. 1
Table of Contents
Introduction .............................................................................................................................................................................................................. 1 Network Topology .................................................................................................................................................................................................... 2 Limitations ................................................................................................................................................................................................................ 3 System Components ................................................................................................................................................................................................. 3
Hardware Requirements ...................................................................................................................................................................................... 3 Software Requirements ....................................................................................................................................................................................... 4
Features .................................................................................................................................................................................................................... 4 Features Supported .............................................................................................................................................................................................. 4 Features Not Supported ....................................................................................................................................................................................... 4
Configuration ............................................................................................................................................................................................................ 5 Configuration Sequence and Tasks ..................................................................................................................................................................... 5 Configuring the Avaya S8500 ............................................................................................................................................................................. 6 Configuring the Cisco Unified Communications Manager ............................................................................................................................... 19
Acronyms ............................................................................................................................................................................................................... 43
Introduction
Avaya Communication Manager 4.0 has introduced TCP support for SIP trunks on the C-LAN. Previously, only TLS was supported; this
enhancement allows for direct SIP trunk connectivity to other vendors’ SIP proxies, without the need for Avaya SIP Enablement Services (SES)
servers. Avaya SES is still required in order to support Avaya SIP end points. Further enhancements are provided with Avaya Communication
Manager version 5.0, including support of SIP Diversion header. This document describes the steps and configurations necessary for Cisco
Unified Communications Manager (Cisco UCM) release 8.5 to interoperate with the Avaya S8500 Communication Manager 5.2.1 using SIP
Early-Offer.
The following items were tested:
Basic call between the two systems and verification of voice path, using both H.323 and digital phones on the Avaya side, and SIP and
SCCP IP phones on the Cisco side. Avaya SIP phones are not tested in this scenario, as SIP trunks between Avaya’s C-LAN circuit
packs and Cisco UCM are typically implemented by customers not wishing to use Avaya SES.
CLIP/CLIR/CNIP/CNIR features: calling party name and number delivery (allowed and restricted).
COLP/CONP/COLR/CONR features: connected name and number delivery (allowed and restricted).
Call transfer: attended, and early attended.
Call forwarding: call forward all (CFA), call forward busy (CFB), and call forward no answer (CFNA).
Hold and resume with music on hold.
Three-way conferencing.
Voice messaging and MWI activation-deactivation.
DTMF-relay via RFC2833.
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Listed below are the highlights of the integration issues:
Basic calls worked from Cisco UCM to Avaya PBX and vice versa. Avaya’s Media Shuffling feature was enabled throughout this testing
exercise.
CLIR/CNIR—The Avaya SIP trunk does not support calling/connected name and number restriction. Restriction of calling number on
Avaya digital and H.323 phones is achieved by configuring the Avaya station configuration page and not the SIP trunk page. This
restriction is honored by Cisco UCM.
COLR/CONR—As with calling name and number presentation restrictions, the Avaya PBX does not support connected name and
number restriction on SIP trunks. Cisco UCM, on the other hand, restricts the connected name and number information the same way as
for calling name and number restriction–by setting the SIP PRIVACY to ―id‖ in the SIP trunk configuration page. Thus, the SIP privacy
setting covers all outgoing message presentation restrictions, whether for inbound or outbound calls.
Both systems support call forwarding (CFU, CFB, and CFNA) features. There are some call forward scenarios where the calling name
and number are not updated after the call has been forwarded. This issue is found primarily when an Avaya phone is either the
originating or terminating end. The Avaya phones (IP or legacy) do not display the forwarding phone’s name and number information,
whether it is a local or network forwarded call. Cisco phones display the forwarding information only when it is a locally forwarded call.
Call completion (callback) feature is not supported on either systems using standard SIP protocol.
Typically, Avaya S8500 PBX uses the History-Info field to send redirecting number information, while the Cisco UCM uses the
Diversion header. This affects how calls are treated when redirected to a voice mail system over an SIP trunk. As of Avaya
Communication Manager releases 5.0, however, Avaya included support of Diversion header. On Avaya PBX’s using older software
releases, Cisco Unified Communications Manager release 8.5 provides the ability to convert Diversion headers into History-Info headers
via SIP Normalization Script. Information on how to configure a SIP Normalization Script converting History-Info headers into
Diversion headers and vice versa is provided within this document.
Below are the key results:
Basic call, call transfer, call forwarding, conference call, and hold and resume work successfully.
DTMF-relay using RFC2833 worked bi-directionally.
Centralized voicemail, using Unity/Unity Connection server integrated to Cisco UCM via SCCP was used for testing. This voicemail
solution can provide centralized voicemail services, supporting both Avaya and Cisco end-users. An Avaya Octel 200 voicemail
platform, integrated via E1 PRI QSIG to the Avaya PBX, was also used for testing. It was determined, however, that Avaya PBX does
not provide SIP-to-QSIG interworking with regards to diversion information: whenever a call forwards from a Cisco end point over SIP
trunk to Avaya Octel 200, no diverting leg information is passed over the D-channel; this limitation causes the voicemail system to
answer with the generic greeting. QSIG MWI to SIP NOTIFY is also not supported by the Avaya PBX.
Network Topology
Network Topology/Testbed Setup
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H.323 Phone B1
x4123
H.323 Phone A1
x4124
Avaya S8500
Communication
Manager 5.2.1
Digital Phone A2
x4151
Digital Phone B2
x4152
Cisco UCM
Release 8.5Cisco Catalyst
6000 Ethernet
Switch
Cisco 7965 SCCP
phones
Dial Plans
4XXX
Cisco 7965 SIP phones
Phone D1
x5014
Phone C1
x5013
SIP Trunk
(TCP)
Dial Plans
5XXX
Phone C2
x5015
Phone D2
x5017
Cisco Unity Connection
Limitations
These are the known limitations, caveats, or integration issues:
Centralized Avaya voicemail using QSIG integration to the Avaya PBX is not supported. SIP-to-QSIG interworking on the Avaya does
not provide diversion information over the QSIG call leg. Centralized voicemail using Cisco Unity/Unity Connection integrated to Cisco
UCM is supported, so long as Diversion header is passed to Cisco Unified Communications Manager. This can be achieved by enabling
support of Diversion Header on the Avaya Communication Manager 5.0 SIP trunk group configuration form, or by using a SIP
Normalization Script converting History-info headers into Diversion headers.
No support for callback across SIP trunk.
System Components
Hardware Requirements
The following hardware is required:
Cisco Unified Communications Manager (Cisco UCM) MCS -7835 server
Cisco Unity Connection MCS-7835 server
Catalyst switch 6506 with WS-X6K-Sup1A-2GE and WS-X6348
Cisco 7960, 7965, 7970, and 7971 IP phones
Avaya S8500 PBX with G650 Media Gateway
Avaya TN799DP C-LAN card
Avaya TN2302AP IP Media Processor card
Avaya 9600-series IP phones and 6400-series digital station phones
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Software Requirements
The following software is required:
Cisco Unified Communications Manager release 8.5
Catalyst 6000 with IOS release: WS-C6006 Software, Version NmpSW: 7.6(8)
Cisco Unity Connection release 8.5
Avaya Communication Manager release 5.2.1
Avaya TN799DP firmware release 39
Avaya TN2302AP firmware release 121
Avaya 9600-series IP phones firmware release 3.1.1 (H.323)
Features
This section lists supported and unsupported features. Please see the Limitations section on page 4 for more information.
Features Supported
CLIP—calling line (number) identification presentation.
CLIR—calling line (number) identification restriction.
CNIP—calling name identification presentation.
CNIR—calling name identification restriction.
Alerting name.
Attended call transfer.
Early attended call transfer.
CFU—call forwarding unconditional.
CFB—call forwarding busy.
CFNA—call forwarding no answer.
COLP—connected line (number) identification presentation.
COLR—connected line (number) identification restriction.
CONP—connected name identification presentation.
CONR—connected name identification restriction.
Hold and resume.
Conference call.
DTMF-relay using RFC2833.
Features Not Supported
Call completion (callback, automatic callback).
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Configuration
Configuration Sequence and Tasks
Avaya S8500 PBX:
1. Configure the IP-Codec-Set, and IP-Network-Region.
2. Configure the IP interface for C-LAN and IP Media Processor cards.
3. Configure Cisco UCM as an IP node-name.
4. Configure the signaling group for the SIP trunk to Cisco UCM.
5. Configure the trunk group for the SIP trunk to Cisco UCM.
6. Configure the H.323 and digital station phone extension.
7. Configure the uniform dialing plan to the Cisco UCM extensions.
8. Configure the route pattern to the Cisco UCM extensions.
Cisco Unified Communications Manager:
1. SIP trunk security profile.
2. Device setting SIP profile.
3. Media resource group and media resource group list.
4. Partitions and calling search space.
5. Assign media resource group list (MRGL) in the default device pool.
6. SIP trunk to the Avaya S8500 PBX.
7. SIP Trunk Normalization Script
8. SIP and SCCP phones device configuration.
9. Route pattern to the Avaya S8500 PBX. 10. CallManager Service Parameter “Duplex Streaming Enabled” set to “True”.
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Configuring the Avaya S8500
Avaya S8500 Software Version and Hardware Configuration List
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Avaya IP Nodes
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Avaya Signaling Group for the SIP Trunk to Cisco UCM
Avaya SIP Trunk Group to Cisco UCM
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Avaya IP-Codec-Set
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Avaya IP-Network-Region
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Avaya S8500 Uniform Dialplan to Cisco UCM extensions (5009-5019)
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Avaya S8500 AAR Analysis (204)
Avaya S8500 Route Pattern 44
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Avaya S8500 H.323 Station 4124 (9600-series IP phone)
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Avaya S8500 Digital station 4151 (6400-series digital phone)
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Configuring the Cisco Unified Communications Manager
Cisco Unified Communications Manager Software Version
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Cisco Unified Communications Manager SIP Trunk Security Profile
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Cisco Unified Communications Manager SIP Trunk Profile
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Cisco Unified Communications Manager SIP Trunk to Avaya S8500 PBX
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Cisco Unified Communications Manager SIP Trunk Normalization Script
Note: Avaya supports SIP History-Info header when providing call forward (diversion) information over SIP trunks. Cisco
Unified Communications Manager normally uses Diversion header. Because of this, a SIP Normalization script is required in
order to convert History-Info headers into Diversion headers. This is useful whenever Cisco Unity/Unity Connection centralized
voicemail (integrated with Cisco Unified Communications Manager) is used to support both Avaya and Cisco end users. The full
content of the SIP Normalization Script is captured below:
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M = {} M.allowHeaders = {"History-Info"} trace.enable() function M.outbound_INVITE(msg) local callid = msg:getHeader("Call-ID") trace.format("M.outbound_INVITE: callid is '%s'", callid) local di = msg:getHeader("Diversion") if not di then return end msg:convertDiversionToHI() msg:removeHeader("Diversion") local historyInfos = msg:getHeaderValues("History-Info") msg:removeHeader("History-Info") local newHi = "" for i, hi in ipairs(historyInfos) do local main_header = string.match(hi, '(.*)?') or string.match(hi, "(.*)>;index=(.*)") local embed_header = string.match (hi, '?Reason=sip(.*)>') local index = string.match(hi, '>;index=(.*)') local hiNext = historyInfos[i + 1] local indexNext = string.match(hiNext or "", '>;index=(.*)') trace.format("main_header is '%s'", main_header or "nil") if i == 1 then local firstHi = string.format("%s>;index=%s", main_header, index) firstHi = string.gsub(firstHi, "@(.*):%d+", "@%1") msg:addHeader("History-Info", firstHi) end if embed_header then trace.format("embed_header is '%s'", embed_header) embed_header = string.gsub(embed_header, "unconditional", "Moved Temporarily") embed_header = string.gsub(embed_header, ";", "%%3B") embed_header = string.gsub(embed_header , "=", "%%3D") embed_header = string.gsub(embed_header, "\"", "%%22") embed_header = string.gsub(embed_header, " ", "%%20") embed_header = string.format("?Reason=SIP%s%s", embed_header, "&Reason=Redirection%3Bcause%3DCFI") end -- Get rid of the port number main_header = string.gsub(main_header, "@(.*):%d+", "@%1")
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if not indexNext then local left, right = string.match(index, "(%d+)%.(%d+)") indexNext = string.format("%s.%s", left + 1, right) end hi = string.format("%s%s>;index=%s", main_header, embed_header or "", indexNext) msg:addHeader("History-Info", hi) end end local HiCauseToDiversion = { } HiCauseToDiversion["302"] = "unconditional" HiCauseToDiversion["486"] = "user-busy" HiCauseToDiversion["408"] = "no-answer" HiCauseToDiversion["480"] = "deflection" HiCauseToDiversion["487"] = "deflection" HiCauseToDiversion["503"] = "unavailable" HiCauseToDiversion["404"] = "unknown" function convertHIToDiversion(msg) local historyInfos = msg:getHeaderValues("History-Info") for i, hi in ipairs(historyInfos) do hi = string.gsub(hi, "%%3B", ";") hi = string.gsub(hi, "%%3D", "=") hi = string.gsub(hi, "%%22", "\"") hi = string.gsub(hi, "%%20", " ") -- Reason=SIP;cause=302;text="Moved Temporarily" local uri, reason, cause, text = string.match(hi, "<(sip:.*@.*)?Reason=(SIP);cause=(%d+);text=(\".*\")") trace.format("hi: uri '%s', reason '%s', cause '%s', text '%s'", uri or "nil", reason or "nil", cause or "nil", text or "nil") if reason == "SIP" then local dReason = HiCauseToDiversion[cause] or "unknown" local diversion = string.format("<%s>;reason=\"%s\"", uri, dReason) msg:addHeader("Diversion", diversion) end end end function M.inbound_INVITE(msg) local callid = msg:getHeader("Call-ID") trace.format("M.inbound_INVITE: callid is '%s'", callid) local hist = msg:getHeader("History-Info") local di = msg:getHeader("Diversion") if hist then local context = msg:getContext() if context then
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context["History-Info"] = hist end if not di then convertHIToDiversion(msg) end end local di = msg:getHeader("Diversion") if di then trace.format(" -- found Diversion header") msg:removeHeader("History-Info") -- replace unknown to unconditional di = string.gsub(di, "unknown", "unconditional") msg:modifyHeader("Diversion", di) end end --[[ function M.outbound_ANY_INVITE(msg) local context = msg:getContext() if context then msg:addHeader("History-Info", context["History-Info"]) end end --]] return M
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Cisco Unified Communications Manager Service Parameter
Note: Cisco Unified Communications Manager Service Parameter ―Duplex Streaming Enabled‖ must be set to ―True‖ in order for
MoH and ringback to work properly during call transfers/conferences initiated by Cisco stations to Avaya IP endpoints.
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Cisco Unified Communications Manager Media Resource Group and Media Resource Group List
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Cisco Unified Communications Manager Route Pattern to Avaya S8500 Configuration
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Cisco Unified Communications Manager SIP Phone Ext. 5015 Device Level Configuration
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Cisco Unified Communications Manager SCCP Phone Ext. 5013 Device Level Configuration
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Acronyms
Acronym Definition
CCBS Call Completion to Busy Subscriber
CCNR Call Completion on No Reply
CFB Call Forwarding on Busy
CFNR Call Forwarding No Reply
CFU Call Forwarding Unconditional
CLIP Calling Line (Number) Identification Presentation
CLIR Calling Line (Number) Identification Restriction
CNIP Calling Name Identification Presentation
CNIR Calling Name Identification Restriction
COLP Connected Line (Number) Identification Presentation
COLR Connected Line (Number) Identification Restriction
CONP Connected Name Identification Presentation
CONR Connected Name Identification Restriction
CT Call Transfer
Cisco UCM Cisco Unified Communications Manager
DNS Domain Name Server
FQDN Fully Qualified Domain Name
MWI Message Waiting Indicator
MRGL Media Resource Group List
MTP Media Termination Point
PSTN Public Switched Telephone Network
SIP Session Initiated Protocol
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Important Information
THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE
WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO
BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE
FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.
IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR
INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA
ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN
ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
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