Avaya S8500 Rel. 5.2.1 using SIP via Cisco Unified

© 2012 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 92 EDCS-1203965 Rev. 1

Application Note

Avaya S8500 Rel. 5.2.1 using SIP via Cisco Unified

Communications Manager–Session Manager Edition 9.0 to Cisco

Unified Communications Manager 9.0 and Cisco Unified Border

Element (Enterprise Edition) Release 9.0 on ISR G2 to Service

Provider

© 2008 Cisco Systems, Inc. All rights reserved.

Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com Page 2 of 92

October 26, 2012 – Rev. 1

Table of Contents

Introduction .............................................................................................................................................................................................................. 4 Network Topology .................................................................................................................................................................................................... 5 Limitations ................................................................................................................................................................................................................ 5 System Components ................................................................................................................................................................................................. 6

Hardware Requirements ...................................................................................................................................................................................... 6 Software Requirements ....................................................................................................................................................................................... 6

Features .................................................................................................................................................................................................................... 7 Features Supported .............................................................................................................................................................................................. 7 Features Not Supported ....................................................................................................................................................................................... 7

Configuration ............................................................................................................................................................................................................ 8 Configuring the Avaya S8500 PBX .................................................................................................................................................................... 8

Software/Firmware Versions .......................................................................................................................................................................... 8 System Parameters IP Options ....................................................................................................................................................................... 9 IP Nodes ....................................................................................................................................................................................................... 11 IP Network Region ....................................................................................................................................................................................... 12 IP Codec Set ................................................................................................................................................................................................. 14 Signaling Group ........................................................................................................................................................................................... 15 Trunk Group ................................................................................................................................................................................................. 15 Route Pattern ................................................................................................................................................................................................ 18 AAR/ARS Analysis ..................................................................................................................................................................................... 19 Uniform Dialing Plan ................................................................................................................................................................................... 20 ISDN Public/Unknown Numbering Plan ..................................................................................................................................................... 21 Incoming-call-handling-trmt ........................................................................................................................................................................ 22 Station Configuration (IP Phone) ................................................................................................................................................................. 23 Station Configuration (Analog Line)............................................................................................................................................................ 25

Configuring the Cisco Unified Communications Manager – Session Manager Edition .................................................................................... 27 Cisco Unified Communications Manager – Session Manager Edition software version ............................................................................. 27 Configuration of Device Pool to Region mapping ....................................................................................................................................... 28 Configuration of Partitions ........................................................................................................................................................................... 28 Configuration of Calling Search Spaces ....................................................................................................................................................... 29 Configuration of Translation Pattern used to strip leading digits on inbound calls from SP ........................................................................ 31 Configuration of SIP Profile used by SIP trunks .......................................................................................................................................... 32 Configuration of SIP Normalization Script (used by SIP trunk to Avaya PBX) .......................................................................................... 38 Configuration of SIP trunks to PSTN ........................................................................................................................................................... 41 Configuration of SIP trunk to Avaya PBX ................................................................................................................................................... 44 Configuration of SIP trunk to Cisco UCM ................................................................................................................................................... 47 Configuration of Route Patterns – To Avaya PBX ...................................................................................................................................... 50 Configuration of Route Patterns – To Cisco UCM ...................................................................................................................................... 52 Configuration of Route Patterns – To PSTN ................................................................................................................................................ 54

Configuring the Cisco Unified Communications Manager ............................................................................................................................... 57 Cisco Unified Communications Manager Software Version ........................................................................................................................ 57 Configuration of Service Parameters – Cisco CallManager ......................................................................................................................... 58 Configuration of Audio Codec Preference List ............................................................................................................................................ 58 Configuration of Device Pool to Region mapping ....................................................................................................................................... 59 Configuration of Conference Bridge ............................................................................................................................................................ 60 Configuration of Media Resource Group ..................................................................................................................................................... 61 Configuration of Media Resource Group List .............................................................................................................................................. 62 Configuration of SIP Profile ........................................................................................................................................................................ 63 Configuration of SIP Trunk to SME ............................................................................................................................................................ 65 Configuration of Route Pattern to PSTN through SME ............................................................................................................................... 68 Configuration of Route Pattern to Avaya PBX through SME ...................................................................................................................... 69 Configuration of Cisco 7965 SIP Phone ...................................................................................................................................................... 70 Configuration of Cisco 7965 SCCP Phone................................................................................................................................................... 75



Configuration of MGCP FAX Gateway (VG224) ........................................................................................................................................ 80 Configuration of MGCP FAX Gateway Analog Endpoint ........................................................................................................................... 81

Configuring the Cisco UBE - Enterprise ........................................................................................................................................................... 82 Acronyms .......................................................................................................................................................................................................... 90 Important Information ....................................................................................................................................................................................... 91



Introduction

This application note describes the necessary steps and configurations for connectivity between Avaya S8500 release 5.2.1, and a Cisco

Unified Communications Manager (Cisco UCM) version 9.0 with Cisco Unified Communications Manager-Session Management

Edition (Cisco UCM-SME) Version 9.0.

The network topology diagram (Figures 1) shows the test setup for end-to-end interoperability between Cisco Unified Communications

Manager (Cisco UCM) Release 9.0 connected to the Avaya S8500 PBX via a Cisco UCM-SME using SIP trunks (between Cisco UCM-

SME and Avaya PBX) and SIP trunks (between the Cisco UCM-SME and Cisco UCM). Features tested are basic call, 3-way (ad-hoc)

conference, call transfer (attended and unattended), call forward (all, busy and no answer), hold/resume, fax transmission, and DTMF

interworking. This test setup also includes a connection to a Service Provider, using SIP trunks. Cisco Unified Border Element (Cisco

UBE) on ISR is used as a session border controller (SBC), providing demarcation, security, and interworking services between the

customer’s private network and the service provider’s SIP network.

During testing, a Cisco ISR 2921 voice gateway was used to run the Cisco Unified Border Element features set. However other Cisco

voice gateways can be used. The decision to choose the Cisco gateway model is left to the customer. The customer should choose a

Cisco IOS gateway model based on the capabilities and the capacity that will be required based on the planned network deployment.

Here is a list of Cisco IOS products capable of running Cisco UBE.

Cisco 3900 Series Integrated Services Routers

Cisco 2900 Series Integrated Services Routers

Cisco AS5350XM Universal Gateway

Cisco AS5400XM Universal Gateway

Cisco ASR 1000 Series Aggregation Services Routers

Consult your Cisco representative for the correct IOS image and for the specific application and Device Unit License and Feature

License requirements for all your Cisco Unified Communications Manager with Cisco Unified Border Element components.

If additional guidance on the Cisco UBE is needed, please refer to the Cisco UBE section on the Cisco Interoperability Portal

(www.cisco.com/go/interoperability).

This configuration was tested using a generic SIP Trunk Service Provider. Results may vary based on Service Provider being used.

http://www.cisco.com/en/US/products/ps10536/index.html




http://www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6790/gatecont/ps5640/data_sheet_c78_57025.html

http://www.cisco.com/go/interoperability



Network Topology

Figure 1. Basic Call Setup

Service Provider

or Simulated Service

Provider CUBECUBE

Avaya S8500

Phone PBX1

Phone PBX2

Cisco UCM-SME

Rel. 9.0(1)

Phone C2

SIP

SIP

SIP

SIP

Cisco UBE (on ISR 2921)

8.8 = IOS 15.2(4)M1

Phone C1

CPE to CPE

CPE to PSTN (SP)

FAX PBX3

FAX C3

Phone SP1 Phone SP1 FAX SP3

Cisco ISR – for IOS

MTP requirement

Cisco UCM

Rel. 9.0(1)

Capabilities

Voice/fax calls including supplementary services can be successfully established between endpoints controlled by the Avaya PBX and

endpoints controlled by the Cisco Unified Communications Manager.

Voice/fax calls including supplementary services can be successfully established between endpoints controlled by the Avaya PBX and

the PSTN, using Cisco UBE as a session border controller.

Limitations

Avaya PBX

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.

T.38 fax-relay Error Correction Mode (ECM). The Avaya PBX does not currently support Error Correction Mode (ECM) during T.38

fax transmissions.



Fax transmissions using Super Group 3 (SG3) fax protocol. The Avaya PBX currently supports fax transmission speeds up to 9600 bps

when T.38 fax relay is used. Also noticed during testing is the PBX’s inability to always detect SG3 fax tones. Because of this,

customers should avoid configuring fax machines connected to the Avaya PBX for SG3 protocol.

Cisco UBE (Enterprise Edition)

None found

Cisco UCM

Cisco UCM does not natively support History-info headers. A SIP Normalization script must be applied to SIP trunk(s) to Avaya PBX in

order to convert History-info (supported the Avaya PBX) to Diversion header (Cisco UCM-supported). This is required whenever a

Cisco UCM-hosted centralized voicemail platforms (such as Unity Connection) is used.

System Components

Hardware Requirements

Cisco MCS 7800 Unified Communications Manager Appliances

2 Cisco Unified IP phone 7965 configured as SCCP phones

2 Cisco Unified IP phone 7975 configured as SIP phones

Avaya S8500 PBX

Avaya digital and IP stations

TN799DP C-LAN Circuit Pack

TN2302AP Med-Pro Circuit Pack

Cisco ISR 2921 (Cisco UBE-Enterprise)

VG224 Analog Gateway (Fax application). Alternatively, other Cisco IOS gateway(s) can be used. This component may be an H.323,

SIP, or MGCP gateway. The protocol is optional, and the choice is left up to the customer’s network design.

Software Requirements

The following software is required:

Cisco Unified Communications Manager Release 9.0 - Session Manager Edition

Cisco Unified Communications Manager Release 9.0 – Cisco UCM

Cisco Unified Border Element (CISCO UBE) Release 9.0 with IOS version 15.2(4)M1. This configuration was tested with C2900-

universalk9_npe-mz.SPA.152-4.M1; however, this document is also applicable to all IOS 15.2(4)M/CUBE 9.0 and 15.3(2)T/CUBE 9.0

software releases. The documented CISCO UBE configuration can be supported with the following IOS feature sets: UNIVERSAL

Consult your Cisco representative for the correct IOS image and for the specific application and Device Unit License and Feature

License requirements for all your Cisco Unified Communications components. For reference, please follow this link:

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html

Cisco Unity Connection (CUC) Release 8.6. This solution was tested with 8.6.1.20002-109

Cisco VG224 Gateway (IP-TDM) IOS version: 15.1(4)M4

Avaya Aura Communication Manager Release 5.2.1

Avaya 9600 Series H.323 IP Phone firmware version 3.1.1

Avaya TN799DP Firmware Vintage 39

Avaya TN2302AP Firmware Vintage 121

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html



Features

This section lists supported and unsupported features.

Features Supported

Basic calls

CLIP-Calling line (Number) identification presentation

CLIR-Calling line (Number) identification restriction

COLP-Connected line (Number) identification presentation

COLR- Connected line (Number) identification restriction

CNIP-Calling name identification presentation

CNIR-Calling name identification restriction

CONP-Connected name identification presentation

CONR- Connected name identification restriction

Consultation transfer – Local and Network/External

Early Attended transfer – Local and Network/External

Call forward Local – Unconditional, Busy and No reply (See Limitations section for details.)

Call forward Network/External – Unconditional, Busy and No reply (See Limitations section for details.)

DTMF interworking (using RFC 2833 DTMF relay)

MWI—Message Waiting Indicator (lamp ON, lamp OFF) across SIP trunk to Avaya PBX

Fax transmissions (T.38 and G.711 pass-through)

Features Not Supported

Centralized QSIG Voicemail hosted by Avaya PBX.



Configuration

This section contains configuration menus and commands and describes configuration sequences and tasks.

Configuring the Avaya S8500 PBX

1. Configure node-name IP table to include Cisco UCM-SME as a valid IP node.

2. Configure the ip-network-region to assign to the SIP trunk.

3. Configure the ip-codec-set to assign to ip-network-region used by the SIP trunk.

4. Add the new signaling group.

5. Add the new trunk group.

6. Add the new route pattern.

7. Configure AAR/ARS Table entries.

8. Configure Uniform Dialing Plan.

9. Configure ISDN Public/Unknown Numbering Table entry.

10. Configure Incoming Call Handling Treatment for trunk group.

Configuration Menus and Commands for Avaya S8500 PBX

Software/Firmware Versions



System Parameters IP Options





IP Nodes



IP Network Region





IP Codec Set

Note: The ip-codec-set configuration above is assigned to SIP trunk(s) using codecs G.729a, G.711Mulaw, and T.38 fax relay. If the

Service Provider does not support T.38 fax relay, a trunk using G.711 codec is required, with FAX Mode set to “off”.



Signaling Group

Trunk Group







Route Pattern

Note: The Route Pattern configuration above is assigned to AAR code 217, used to route calls to Cisco UCM’s 4-digit extensions. As you

will notice, route pattern 445 “strips” the first 3 digits of the routing number (AAR code 217) before the INVITE message is transmitted

over the SIP trunk.



Note: The Route Pattern configuration above is assigned to ARS analysis table entry 1408, used to route calls to the SP using 1+10-digit

dialing. As you will notice, route pattern 446 does not perform any “digit stripping” before the INVITE message is transmitted over the

SIP trunk, thus using all 11 digits.

AAR/ARS Analysis

Note: AAR code 217 is used to route calls to Cisco UCM extensions over Route Pattern 445. This code will be used in the Uniform

Dialplan table, and will be prefixed onto the 4-digit numbers assigned to Cisco UCM stations. As stated previously, Route Pattern 445

will “strip” the AAR code, leaving only the dialed 4-digit number in the outbound INVITE message.



Note: Dial String 1408 is used to route calls to the Service Provider. After dialing 9 (ARS access code) plus 1-408-nxx-xxxx, all 11 digits

are included in the outbound INVITE message (parameter “Call Type” must be set to “natl”) and the call is routed over Route Pattern

446. This Route Pattern is configured not to “strip” any leading digits.

Uniform Dialing Plan



ISDN Public/Unknown Numbering Plan

Note: The table above is used to define numbering plans to be used on ISDN/SIP calls. In the example above, 4-digit extensions in the

5XXX range are used on Cisco UCM, while 4-digit extensions in the 4XXX range are used by the Avaya PBX. Also note the digit

transformation performed for extensions 4114 and 4124: when calls are sent over trunk group 445 (trunk group used to connect the

PBX to Cisco UCM-SME), ext. 4114 is “transformed” into phone number 732-320-4353, while ext. 4124 is “transformed” into phone

number 732-320-4353 (SIP trunk Service Provider DID numbers).



Incoming-call-handling-trmt

Note: The table above is used to apply changes to incoming called numbers. In this case, DID numbers provided by the Service Provider

do not match extension numbers, and have to be “translated”. SP-provided DID phone numbers 732-320-4352 and 732-320-4353, after

being modified by Cisco UCM-SME (configured to strip all leading digits, and pass the last 4 digits over the next call leg), need to also

be changed to their assigned extension numbers. As you will notice, this table is configured to change number 4352 to ext. 4124, and

4353 to ext. 4124



Station Configuration (IP Phone)





Station Configuration (Analog Line)





Configuring the Cisco Unified Communications Manager – Session Manager Edition

1. Cisco UCM - Session Manager Edition software version

2. Device Pool and Region mapping configuration

3. Partitions configuration

4. Calling Search Space configuration

5. Translation Pattern configuration

6. SIP profile (used by SIP trunks) configuration

7. SIP Normalization Script (used by SIP trunk to Avaya PBX) configuration

8. SIP trunk configuration to SP

9. SIP trunk configuration to Avaya PBX

10. SIP Trunk configuration to Cisco UCM

11. Route Pattern configuration to Avaya PBX

12. Route Pattern configuration to Cisco UCM

13. Route Pattern configuration to SP

Cisco Unified Communications Manager – Session Manager Edition software version



Configuration of Device Pool to Region mapping

Navigation Path: System Region

Configuration of Partitions

Navigation Path: Call Routing Class of Control Partition



Configuration of Calling Search Spaces

Navigation Path: Call Routing Class of Control Calling Search Space





Configuration of Translation Pattern used to strip leading digits on inbound calls from SP

Navigation Path: Call Routing Translation Pattern

Note: The Translation Pattern above is used by Cisco UCM - SME in order to strip leading digits from the called party number on

inbound calls. All leading digits are stripped, and only the last 4 digits are passed to the Avaya PBX and Cisco UCM. As you’ll notice

above, the SIP Trunk Service Provider prefixes 10-digit DID numbers with 5 zero’s.



Configuration of SIP Profile used by SIP trunks

Navigation Path: Device Device Settings SIP Profile













Configuration of SIP Normalization Script (used by SIP trunk to Avaya PBX)

Navigation Path: Device Device Settings SIP Normalization Script

Note: Historically, Avaya supported SIP History-Info header when providing call forward (diversion) information over SIP trunks. As

of Avaya Communication Manager software version 5.X, however, SIP trunk groups can be configured to provide Diversion header.

Cisco Unified Communications Manager uses Diversion header. A SIP Normalization script can be used in order to convert History-

Info headers into Diversion headers when connecting to Avaya PBX’s older than software version CM 5.X. 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

Configuration of SIP trunks to PSTN

Navigation Path: Device Trunk





Note: Generally, SIP Trunk Service Providers require all outbound calls over SIP trunk to provide a valid 10-digit DID number as

caller ID. Configuring a valid 10-digit Caller ID DN in parameter Caller Information Caller ID DN ensures that outbound calls will

be processed even when originating from telephones not configured with valid 10-digit DID numbers.



Configuration of SIP trunk to Avaya PBX








Configuration of SIP trunk to Cisco UCM








Configuration of Route Patterns – To Avaya PBX

Navigation Path: Call Routing Route/Hunt Route Pattern





Configuration of Route Patterns – To Cisco UCM






Configuration of Route Patterns – To PSTN








Configuring the Cisco Unified Communications Manager

1. Cisco Unified Communications Manager Version

2. Service Parameters configuration

3. Audio Codec Preference List configuration

4. Device pool and Region mapping configuration

5. Conference Bridge configuration

6. Media Resource Group configuration

7. Media Resource Group List configuration

8. SIP Profile configuration

9. SIP Trunk to SME configuration

10. Route Pattern configuration to SP

11. Route Pattern configuration to Avaya

12. Cisco IP Phone 7960 SCCP Configuration

13. Cisco IP Phone 7960 SIP Configuration

14. MGCP Fax gateway configuration

Cisco Unified Communications Manager Software Version



Configuration of Service Parameters – Cisco CallManager

Navigation Path: System Service Parameters

Note: Service Parameter “Duplex Streaming Enabled” must be set to “True” in order to successfully provide MoH/Ringback to Avaya

IP phones (H.323) and outside (PSTN) callers when calls are placed on hold and/or transferred from Cisco UCM stations.

Configuration of Audio Codec Preference List

Navigation Path: System Region Information Audio Codec Preference List



Note: CISCO UCM 9.0 introduced a new feature: Audio Codec Preference List. This feature allows for configuration of the order of

audio codec preference, both for inter- and intra-Region calls. Add a new Audio Codec Preference List, with G.729 codecs configured

above G.711 (higher priority). This new Audio Codec Preference list is assigned to the Region used by the Device Pool for Conference

Bridges, Transcoders, and phones requiring G.711 codec. With this configuration in place, inbound calls, as well as call conferences

initiated by Cisco IP phones, will use G.729 codec as their first choice codec.

Configuration of Device Pool to Region mapping

Navigation Path: System Region Information Region



Configuration of Conference Bridge

Navigation Path: Media Resources Conference Bridge



Conference Bridge IOS configuration:

sccp local GigabitEthernet0/0

sccp ccm 172.20.236.50 identifier 1 version 7.0

sccp

!

sccp ccm group 1

bind interface GigabitEthernet0/0

associate ccm 1

priority 1

associate profile 98 register cfb112233445566

!

dspfarm profile 98 conference

codec g729r8

codec g711ulaw

maximum sessions 8

associate application SCCP

Configuration of Media Resource Group

Navigation Path: Media Resources Media Resource Group



Configuration of Media Resource Group List

Navigation Path: Media Resources Media Resource Group List



Configuration of SIP Profile

Navigation Path: Device Device Settings SIP Profile





Configuration of SIP Trunk to SME








Configuration of Route Pattern to PSTN through SME




Configuration of Route Pattern to Avaya PBX through SME




Configuration of Cisco 7965 SIP Phone

Navigation Path: Device Phone











Configuration of Cisco 7965 SCCP Phone

Navigation Path: Device Phone











Configuration of MGCP FAX Gateway (VG224)

Navigation Path: Device Gateway



Configuration of MGCP FAX Gateway Analog Endpoint

Navigation Path: Device Gateway



Configuring the Cisco UBE - Enterprise

CUBE_ISRG2_ATT#show version

Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9_NPE-M), Version 15.2(4)M1,

RELEASE SOFTWARE (fc1)

Technical Support: http://www.cisco.com/techsupport

Copyright (c) 1986-2012 by Cisco Systems, Inc.

Compiled Thu 26-Jul-12 20:54 by prod_rel_team

ROM: System Bootstrap, Version 15.0(1r)M1, RELEASE SOFTWARE (fc1)

CUBE_ISRG2_ATT uptime is 1 week, 5 days, 6 hours, 0 minutes

System returned to ROM by power-on

System image file is "flash:c2900-universalk9_npe-mz.SPA.152-4.M1.bin"

Last reload type: Normal Reload

Last reload reason: power-on

Cisco CISCO2921/K9 (revision 1.0) with 487424K/36864K bytes of memory.

Processor board ID FTX1348AHMN

3 Gigabit Ethernet interfaces

1 terminal line

DRAM configuration is 64 bits wide with parity enabled.

255K bytes of non-volatile configuration memory.

255488K bytes of ATA System CompactFlash 0 (Read/Write)

License Info:

License UDI:

-------------------------------------------------

Device# PID SN

-------------------------------------------------

*0 CISCO2921/K9 FTX1348AHMN

Technology Package License Information for Module:'c2900'

------------------------------------------------------------------------------------

Technology Technology-package Technology-package

Current Type Next reboot

------------------------------------------------------------------------------------

ipbase ipbasek9 Permanent ipbasek9

security None None None

uc uck9 Permanent uck9

data None None None

Configuration register is 0x2102

CUBE_ISRG2_ATT#show running

Building configuration...

Current configuration : 10206 bytes

!

! Last configuration change at 10:45:48 PST Mon Aug 27 2012

version 15.2

service timestamps debug datetime msec

service timestamps log datetime msec



no service password-encryption

service sequence-numbers

!

hostname CUBE_ISRG2_ATT

!

boot-start-marker

boot system flash:c2900-universalk9_npe-mz.SPA.152-4.M1.bin

boot-end-marker

!

!

logging queue-limit 10000

logging buffered 20000000

logging persistent filesize 20000000

logging rate-limit 10000

no logging console

enable secret 5 $1$4jgu$npJCRdswNO47pZhBy3fbi/

enable password cisco

!

no aaa new-model

clock timezone PST -8 0

!

!

!

ip domain name yourdomain.com

no ip cef

no ipv6 cef

multilink bundle-name authenticated

!

!

!

crypto pki trustpoint TP-self-signed-1100168695

enrollment selfsigned

subject-name cn=IOS-Self-Signed-Certificate-1100168695

revocation-check none

rsakeypair TP-self-signed-1100168695

!

!

crypto pki certificate chain TP-self-signed-1100168695

certificate self-signed 01

30820254 308201BD A0030201 02020101 300D0609 2A864886 F70D0101 04050030

31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274

69666963 6174652D 31313030 31363836 3935301E 170D3132 30383038 31363134

33365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649

4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 31303031

36383639 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281

8100DC2A 57584BA4 9E848CCB AE8FC0F9 4CC9D68A 56FEE2D8 B3711442 0A45DCE7

875F5622 E504025D 3BF6DDE7 894E8DDC 6F7A6E21 7EB2012F 551B8100 DBF61436

89CB7CA4 2C40EDCD 395AC7D0 F759F0C8 E942220B FDB9F6E9 34067A81 DE1BE5A9

DA7FB98E 533FB54E E3747FA3 758F19D9 BA886C9A 16FCD1A7 3B3DD80C 195110B2

A62B0203 010001A3 7C307A30 0F060355 1D130101 FF040530 030101FF 30270603

551D1104 20301E82 1C55434D 2D495352 47322D41 54542E79 6F757264 6F6D6169

6E2E636F 6D301F06 03551D23 04183016 80148994 E1E21AE2 848D4387 C647F727

5DB0345F 0ACC301D 0603551D 0E041604 148994E1 E21AE284 8D4387C6 47F7275D

B0345F0A CC300D06 092A8648 86F70D01 01040500 03818100 050B77EA 063D97C6

75C7E8E6 C256D569 86872149 ABD193ED DAE3220D F41FA5F0 867494BD D53F54A0

0BCD8990 2108F2E0 74D872AA 7038638F A1316A6A ADC28EDA 440B4483 DB11E722

F695EA40 98762C08 34FF30F9 0ADF2F99 516B51BF F285B76D 4A047C85 04DA0632

B55DCDF7 000F9BAC 22E7899F 50C2E66C 7A2CCC7E 7A82FEA6

quit



voice-card 0

dspfarm

dsp services dspfarm

!

!

!

voice service voip

address-hiding

mode border-element

allow-connections sip to sip

no supplementary-service sip moved-temporarily

redirect ip2ip

h323

sip

error-passthru

asserted-id pai1 early-offer forced

midcall-signaling passthru

privacy-policy passthru

g729 annexb-all

!

voice class codec 12 codec preference 1 g729r8 bytes 30

codec preference 2 g711ulaw

!

voice class sip-profiles 13 response ANY sip-header Allow-Header modify "UPDATE," ""

request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"

response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"

request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""

request INVITE sdp-header Audio-Attribute add "a=ptime:30"

!

!

license udi pid CISCO2921/K9 sn FTX1348AHMN

!

!

archive

log config

hidekeys

username Cisco password 0 cisco

!

redundancy

!

!

no ip ftp passive

!

translation-rule 1

!

!

!

1 This command enables router to send P-Asserted ID within the SIP Message Header. Alternatively, this command can also be applied to

individual dial-peers (voice-class sip asserted-id pai) 2 This command configures the codec preference to be assigned to dial-peers. Alternatively, single codec’s can be configured into individual

dial-peers 3 This SIP Profile removes UPDATE from SIP Message Header to/from Cisco UCM, as it can cause problems during unattended call transfers.

By default, Cisco 6900-series IP phones use ptime value of 20 ms. Some Service Provider networks prefer ptime value of 30 ms. This SIP

profile modifies SDP ptime value from 20 to 30. It is also used to remove SDP media attribute "a=T38FaxFillBitRemoval:0", as it can cause

T.38 fax relay transmissions to fail



!

!

interface Embedded-Service-Engine0/0

no ip address

shutdown

!

interface GigabitEthernet0/0

description Connection to UC Interop lab network

ip address 172.20.110.158 255.255.255.0

duplex auto

speed auto

!


description connection to ATT Network

ip address 99.136.XXX.XXX 255.255.255.0

duplex auto

speed auto

!


no ip address

shutdown

duplex auto

speed auto

!

ip default-gateway 172.20.110.1

ip forward-protocol nd

!

ip http server

ip http access-class 23

ip http authentication local

ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

!

ip dns server view-group DNS

no ip pim dm-fallback

ip route 172.20.0.0 255.255.0.0 172.20.110.1

ip route 172.30.0.0 255.255.0.0 172.20.43.1

ip route 207.242.XXX.XXX 255.255.255.255 99.136.XXX.XXX

ip route 207.242.XXX.XXX 255.255.255.255 99.136.XXX.XXX

!

access-list 23 permit 10.10.10.0 0.0.0.7

!

!

control-plane

!

!

no ccm-manager fax protocol cisco

ccm-manager music-on-hold

!

mgcp

no mgcp package-capability res-package

no mgcp package-capability fxr-package

mgcp fax t38 ecm

!

mgcp profile default

!

!

dial-peer voice 1999 voip

description Outgoing calls to SP - Facing SP network



destination-pattern 1T

session protocol sipv2

session target ipv4:207.242.xxx.xxx

voice-class codec 1

voice-class sip asymmetric payload full

voice-class sip asserted-id pai

voice-class sip privacy-policy passthru

voice-class sip early-offer forced

voice-class sip profiles 1

dtmf-relay rtp-nte4 fax-relay sg3-to-g3

fax rate 14400 bytes 48

fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw5 no vad

!


description Outgoing calls to SP - Facing Cisco UCM-SME


incoming called-number 1T

voice-class codec 1




dtmf-relay rtp-nte

fax-relay sg3-to-g3

fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw

no vad

!


description Incoming calls from SP - Facing Cisco UCM-SME

destination-pattern 00000[37][13][24].......


session target ipv4:172.20.236.252

voice-class codec 1





dtmf-relay rtp-nte

fax-relay sg3-to-g3



no vad

!


description Incoming calls from SP - Facing SP network


incoming called-number 00000[37][13][24].......

voice-class codec 1




dtmf-relay rtp-nte

fax-relay sg3-to-g3

4 This command enables DTMF digit passing using RTP NTE (RFC2833) to calls matching this dial-peer

5 This command enables Cisco UBE to perform T.38 fax relay. To change fax protocol to pass-through using G.711mulaw, the command has to

be changed to “fax protocol pass-through g711ulaw”



fax rate 14400


no vad

!


description Outgoing Intl Calls to SP - Facing SP network




voice-class codec 1






dtmf-relay rtp-nte

fax-relay sg3-to-g3



no vad

!


description Outgoing Intl calls to SP - Facing Cisco UCM-SME



voice-class codec 1



dtmf-relay rtp-nte

fax-relay sg3-to-g3


no vad

!


description N11 Calls to SP - Facing SP network

destination-pattern [459]11



voice-class codec 1






dtmf-relay rtp-nte

no vad

!


description N11 Calls to SP - Facing Cisco UCM-SME


incoming called-number .11

voice-class codec 1




dtmf-relay rtp-nte

no vad

!




description Incoming local 7-digit calls from SP - Facing Cisco UCM-SME

destination-pattern 3204...


session target ipv4:172.20.236.50

voice-class codec 1






dtmf-relay rtp-nte

fax-relay sg3-to-g3



no vad

!


description Incoming local 7-digit calls from SP - Facing SP network


incoming called-number 3204...

voice-class codec 1




dtmf-relay rtp-nte

fax-relay sg3-to-g3



no vad

!


description Outgoing call to Operator – Facing SP network




voice-class codec 1






dtmf-relay rtp-nte

no vad

!


description Outgoing call to Operator – Facing Cisco UCM-SME



voice-class codec 1




dtmf-relay rtp-nte

no vad

!

!

sip-ua

no remote-party-id



disable-early-media 1806 retry invite 2

!

!

!

gatekeeper

shutdown

!

!

!

line con 0

exec-timeout 0 0

line aux 0

line 2

no activation-character

no exec

transport preferred none

transport input all

transport output pad telnet rlogin lapb-ta mop udptn v120 ssh

stopbits 1

line vty 0 4

exec-timeout 15 0

password cisco

login

transport input all

!

scheduler allocate 20000 1000

!

end

6 This command allows Cisco UBE to disable early-media upon receiving 180 Ringing message. It is required in order to provide ringback tone

during unattended call transfers



Acronyms

Acronym Definitions

ANF-PR Additional Network Feature Path Replacement

AOC Advice-of-charge. Information element is sent with the connection setup information for incoming Euro-ISDN connections. The AOC IE is used for call charge calculation.

Cisco UCM Cisco Unified Communications Manager

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

MWI Message Waiting Indicator

PSTN Public Switched Telephone Network

SP Service Provider



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 Rel. 5.2.1 using SIP via Cisco Unified