Avaya S8500 Release CM 5.2.1 using SIP trunk to … · Avaya SES is still required in order to support ... for calling name and number restriction–by setting the SIP PRIVACY to

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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.



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

Figure 1. Basic Call Setup



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



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).



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



Configuring the Avaya S8500

Avaya S8500 Software Version and Hardware Configuration List



Avaya IP Nodes



Avaya Signaling Group for the SIP Trunk to Cisco UCM



Avaya SIP Trunk Group to Cisco UCM







Avaya IP-Codec-Set

Setting the codec to G.729 for Audio-Codec-Preference testing.





Avaya IP-Network-Region







Avaya S8500 Uniform Dialplan to Cisco UCM extensions (3000-3005)



Avaya S8500 AAR Analysis (209)



Avaya S8500 Route Pattern 9



Avaya S8500 SIP Station 2003 (9600-series IP phone)



Avaya S8500 Digital station 2151(6400-series digital phone)





The above configuration is for CFwdAll testing (ex. To phone 3000 in this case)





Configuring the Cisco Unified Communications Manager

Cisco Unified Communications Manager Software Version



Cisco Unified Communications Manager SIP Trunk Security Profile



Cisco Unified Communications Manager SIP Trunk Profile









Cisco Unified Communications Manager SIP Trunk to Avaya S8500 PBX







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:



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") 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 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



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.



Cisco Unified Communications Manager Media Resource Group and Media Resource Group List





Cisco Unified Communications Manager Route Pattern to Avaya S8500 Configuration





Cisco Unified Communications Manager SIP Phone Ext. 3005 Device Level Configuration









Cisco Unified Communications Manager SCCP Phone Ext. 3001 Device Level Configuration









Cisco Unified Communications Manager Audio Codec Preference List Configuration









Cisco Unified Communications Manager Region and MRG configuration













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



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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Documents

Avaya S8500 Release CM 5.2.1 using SIP trunk to … · Avaya SES is still required in order to support ... for calling name and number restriction–by setting the SIP PRIVACY to