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EDCS# 1200326 Rev # “Initial Version”
Application Note
Avaya S8500 Release CM 5.2.1 using SIP trunk to Cisco Unified
Communications Manager Release 9.0
October 12, 2012- Initial Version
Table of Contents
Introduction .............................................................................................................................................................................................................. 2 Network Topology .................................................................................................................................................................................................... 4 Limitations ................................................................................................................................................................................................................ 5 System Components ................................................................................................................................................................................................. 5
Hardware Requirements ...................................................................................................................................................................................... 5 Software Requirements ....................................................................................................................................................................................... 5
Features .................................................................................................................................................................................................................... 6 Features Supported .............................................................................................................................................................................................. 6 Features Not Supported ....................................................................................................................................................................................... 6
Configuration ............................................................................................................................................................................................................ 7 Configuring Sequence and Tasks: ....................................................................................................................................................................... 7 Configuring the Avaya S8500 ............................................................................................................................................................................. 8 Configuring the Cisco Unified Communications Manager ............................................................................................................................... 27
Acronyms ............................................................................................................................................................................................................... 63 Important Information ............................................................................................................................................................................................ 64
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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 9.0 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 SIP and digital phones on the Avaya side, and SIP and
SCCP IP phones on the Cisco side.
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.
Audio Codec Preference List – New feature with CUCM 9.0
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 SIP 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.
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 9.0 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.
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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.
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Network Topology
Figure 1. Basic Call Setup
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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.
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 7961, 7971, 7970, and 9971 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
Software Requirements
The following software is required:
Cisco Unified Communications Manager release 9.0
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
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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.
MWI—Message Waiting Indicator (lamp ON, lamp OFF).
Audio Codec Preference List
Features Not Supported
Call completion (callback, automatic callback).
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Configuration
Configuring 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 SIP 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”. 11. Audio Codec Preference Configuration 12. Region Configuration
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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
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Avaya SIP Trunk Group to Cisco UCM
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Avaya IP-Codec-Set
Setting the codec to G.729 for Audio-Codec-Preference testing.
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Avaya IP-Network-Region
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Avaya S8500 Uniform Dialplan to Cisco UCM extensions (3000-3005)
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Avaya S8500 AAR Analysis (209)
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Avaya S8500 Route Pattern 9
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Avaya S8500 SIP Station 2003 (9600-series IP phone)
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Avaya S8500 Digital station 2151(6400-series digital phone)
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The above configuration is for CFwdAll testing (ex. To phone 3000 in this case)
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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: SIP Normalization script is optional and is only required when there is a centralized voicemail platform integrated to the
CUCM, and only when the Avaya PBX in question is pre-CM 5.0 release and cannot be configured to pass diversion header
natively.
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")
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end -- Get rid of the port number main_header = string.gsub(main_header, "@(.*):%d+", "@%1") 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")
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if hist then local context = msg:getContext() if context then 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. 3005 Device Level Configuration
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Cisco Unified Communications Manager SCCP Phone Ext. 3001 Device Level Configuration
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Cisco Unified Communications Manager Audio Codec Preference List Configuration
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Cisco Unified Communications Manager Region and MRG 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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