Avaya S8300 Release 6.0.1 using SIP via Cisco Unified ... · © 2014 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems,

© 2014 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

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Application Note

Avaya S8300 Release 6.0.1 using SIP via Cisco Unified Communications

Manager–Session Manager Edition 10.0 to Cisco Unified Communications

Manager 10.0 and Cisco Unified Border Element (Enterprise Edition) IOS

Release 15.3(3) M on ISR G2 to Service Provider


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Table of Contents

Introduction ............................................................................................................................................ 5

The following items were tested: ........................................................................................................... 5

Network Topology .................................................................................................................................. 6

Limitations .............................................................................................................................................. 7

Capabilities ............................................................................................................................................. 8

System Components ............................................................................................................................... 8

Hardware Requirements ........................................................................................................................ 8

Software Requirements.......................................................................................................................... 8

Features .................................................................................................................................................. 9

Features Supported ................................................................................................................................ 9

Features Not Supported / Not Tested..................................................................................................... 9

Configuration ........................................................................................................................................ 10

Configuring the Avaya S8300 PBX........................................................................................................ 10

System Parameters IP Options ........................................................................................................... 13

IP Nodes ............................................................................................................................................. 15

IP Network Region.............................................................................................................................. 16

IP Codec Set ....................................................................................................................................... 19

Signaling Group .................................................................................................................................. 21

Trunk Group ....................................................................................................................................... 22

Route Pattern ..................................................................................................................................... 26

AAR/ARS Analysis ............................................................................................................................... 28

Uniform Dialing Plan .......................................................................................................................... 30

ISDN Public/Unknown Numbering Plan .............................................................................................. 31

Incoming-call-handling-treatment ...................................................................................................... 32

Station Configuration (IP Phone) ........................................................................................................ 33

Station Configuration (Analog Line - FAX) ........................................................................................... 36


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Configuring the Avaya Aura – Session Manager Edition ...................................................................... 38

Domains: ............................................................................................................................................ 39

Locations: ........................................................................................................................................... 40

Adaptations:....................................................................................................................................... 41

SIP Entities: ........................................................................................................................................ 43

Routing Policies .................................................................................................................................. 47

Dial Pattern to reach CUCM: .............................................................................................................. 48

Configuring the Cisco Unified Communications Manager – Session Manager Edition ........................ 49

Cisco Unified Communications Manager – Session Manager Edition software version ....................... 50

Configuration of Partitions ................................................................................................................. 51

Configuration of Calling Search Spaces ............................................................................................... 52

Configuration of Translation Pattern used to strip leading digits on inbound calls from SP ................. 55

Configuration of SIP Profile used by SIP trunks ................................................................................... 57

SIP Profile with Early Media disabled: ................................................................................................. 61

Configuration of SIP Normalization Script (used by SIP trunk to Avaya PBX) ....................................... 66

Complete Script:................................................................................................................................. 67

Configuration of SIP trunks to PSTN.................................................................................................... 68

Configuration of SIP trunk to Avaya PBX ............................................................................................. 74

Configuration of SIP trunk to Cisco UCM ............................................................................................ 78

Configuration of SIP trunk to Unity Connection .................................................................................. 84

Configuration of Route Patterns – To Avaya PBX ................................................................................ 89

Configuration of Route Patterns – To Cisco UCM ................................................................................ 92

Configuration of Route Patterns – To PSTN ........................................................................................ 94

Configuring the Cisco Unified Communications Manager ................................................................... 98

Cisco Unified Communications Manager Software Version ................................................................ 99

Configuration of Service Parameters – Cisco Call Manager ............................................................... 100

Configuration of Audio Codec Preference List .................................................................................. 101

Configuration of Regions .................................................................................................................. 104


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Configuration of Device Pool ............................................................................................................ 107

Configuration of Conference Bridge ................................................................................................. 114

Configuration of Media Resource Group .......................................................................................... 115

Configuration of Media Resource Group List .................................................................................... 116

Configuration of SIP Profile .............................................................................................................. 117

SIP Profile with Early Media Disabled: .............................................................................................. 121

Configuration of SIP Trunk Security Profile ....................................................................................... 125

Configuration of SIP Trunk to SME .................................................................................................... 126

Configuration of Route Pattern to PSTN through SME ...................................................................... 131

Configuration of Route Pattern to Avaya PBX through SME .............................................................. 133

Configuration of Cisco 9971 SIP Phone ............................................................................................. 135

Configuration of Cisco 8945 SCCP Phone .......................................................................................... 144

Configuration of Cisco Unity Connection User .................................................................................. 152

Configuration of FAX Gateway .......................................................................................................... 155

Configuration of FAX Gateway Analog Endpoint ............................................................................... 157

FAX Gateway Configuration: ............................................................................................................. 159

Configuring the Cisco UBE - Enterprise .............................................................................................. 168

Acronyms ............................................................................................................................................ 177

Test Results ................................................................................................. Error! Bookmark not defined.


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Introduction

This application note describes the necessary steps and configurations for connectivity between Avaya S8300 Communications Manager Release 6.0, and a Cisco Unified Communications Manager (Cisco UCM) Release 10.0 with Cisco Unified Communications Manager-Session Management Edition (Cisco UCM-SME) Release 10.0.

The network topology diagram (Figures 1) shows the test setup for end-to-end interoperability between Cisco Unified Communications Manager (Cisco UCM) Release 10.0 connected to the Avaya S8300 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).

This test setup also includes a connection to a Service Provider, using Verizon 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.

All the test scenarios have been carried out between Avaya (SIP) and Cisco (SIP & SCCP) end points.

The following items were tested:

Basic call (Audio & Video) – Basic calls have been thoroughly tested between Avaya, Cisco & PSTN Endpoints

Calling Name Delivery

Private Number Display

Call Hold-Resume with MOH

Off-net/Intra site Call Conference

IP teleconferencing

Call Transfer (attended and unattended)

Call Forward ( Unconditional, Busy, Don’t answer)

Simultaneous calls

Voice Mail, MWI & Auto-attendant

Dial Plans

Audio Codec Preference list

Fax transmission


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Network Topology

CPE-PBX2

X4501 SIP

Avaya one X SIP

CPE-PBX1

X4500 SIP

Avaya one X

PSTN-

SP2

PSTN-

SP1PSTN-

FAX

SIP

Avaya S8300

Communication

Manager R6.0

Cisco Unity

Connection Release

10.0

Avaya Aura

Session

Manager R6.1

CUCM-Session

Manager Edition

Release 10.0

Cisco Unified

Border Element

Version 15.3

Cisco Unified

Communications

Manager R10.0

CPE-C1

x6002 SIP

Cisco 9971

CPE-C2

x6005 Cisco

9971

CPE_C3

x6010 SCCP

Cisco 8945

CPE-Fax

x4100

Venta Fax

CPE-Fax

X6100

Venta Fax

CPE-C4

x6008 SCCP

Cisco 8945


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Limitations

Avaya PBX did not host Voice Mail server. Centralized Voice Mail using Cisco Unity Connection integrated to Cisco Unified Call Manager – Session Manager Edition (SME) via SIP is used for testing.

The tested version of Avaya PBX supports Diversion header, but versions older than 5.0 do not. The voice mail integration described in this document assumes Avaya support for Diversion header.

Restriction of Calling Number is configured at station level on Avaya PBX since Avaya SIP trunk does not support Calling/Connect name and number restriction.

Call-info header needs to be removed from incoming INVITE to Avaya in order establish successful calls between Cisco and Avaya. This is accomplished via LUA script on the SIP trunk towards Avaya.

Video Call Transfer and Video Conference call failures, appeared to be Avaya is not responding to a re-INVITE sent from CUCM after the redirection, thus there is no audio after the redirection and call drops.

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, for a local forwarded call. Cisco phones display the forwarding information only when it is a locally forwarded call.

T.38 fax from Avaya to Cisco fails to complete if Out of Band DTMF is enabled on Avaya. The work around for this limitation is to disable out of band DTMF on Avaya. A defect ID CSCun32496 has been opened to resolve the issue.

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. PSTN end points also have some issues detecting SG3 tone and hence fax rate is restricted accordingly.

Cisco UBE (Unified Border Element) doesn’t appear to be applying the codec preferences configured. It always seems to be imposing its own codec preferences and ends up establishing the call over G.729.

Feature access codes are not actually supported by the service provider we used for our testing.


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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.

System Components

Hardware Requirements

The following hardware was used:

Cisco UCS-C240-M3S VMWare host running ESXi 5.5 Standard

CUBE- Cisco CISCO2951/K9 (revision 1.1) with 1835264K/261888K bytes of memory with 3 Gigabit Ethernet interfaces

Catalyst switch 3750 WS-C3750X-48

2 Cisco Unified IP phone 9971 configured as SIP phones

2 Cisco Unified IP phone 8945 configured as SIP phones

Avaya S8300D PBX with G6430 Media Gateway

Avaya Common Server HP DL360 G7 with Session Manager

Avaya Common Server HP DL360 G7 with System Manager

Avaya Common Server HP DL360 G7 with Session border Controller

Avaya 4600 series IP telephone (H323)

Software Requirements

The following software is required:

Cisco Unified Communications Manager release 10.0.1.10000-24

Cisco Unity Connection release 10.0.97028-14

Cisco Unified Border Element IOS Version 15.3(3)M, CUBE software release 10.0

Avaya Communication Manager release 6.01 Service Pack 11(patch 20685) (System Platform 6.0.3.10.3)


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Avaya G430 Media Gateway firmware release 30.12.1

Avaya Aura® Session Manager R6.1 (6.1.2.0.612004) Service Pack 2

Avaya Aura® System Manager R6.1 (System Platform 6.0.3.0.3, Template 6.1.5.0) Service Pack 2

Catalyst 3750 Cisco IOS Software, C3750E Software (C3750E-UNIVERSALK9-M), Version 12.2(55)SE5

Avaya Aura® Session Border Controller 6.1 (System Platform 6.0.3.0.3, Template E362P4)

Avaya One-X Communicator Release 6.1

Features

This section lists supported and unsupported features.

Features Supported

Basic calls (Audio & Video)

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 / Not Tested

Centralized Voicemail hosted by Avaya PBX.


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Call completion (callback, automatic callback).

Inter-working Test Cases with Various Calling/Connected Name and Number.

Interworking Test Cases for Call Transfer

Feature Access codes

Configuration

The goal of this guide is to provide an overview of the integration with Cisco Session Manager Edition connected to Verizon SIP trunk via Cisco UBE and two PBX’s Avaya (CM, SM) and Cisco Unified Call Manager.

This deployment used will interconnect UC systems through a single front-end system, in this case the Unified CM Session Management Edition. Unified CM Session Management Edition is essentially a Unified CM cluster with trunk interfaces only and generally no IP endpoints. It enables aggregation of multiple UC systems, referred to as leaf systems.

Typically, Unified CM Session Management Edition is used to interconnect multiple Unified CM clusters, third-party UC systems (IP- and TDM-based PBXs), PSTN connections, and centralized UC applications.

Cisco recommends deploying Unified CM Session Management Edition to do any of the following:

Create and manage a centralized dial plan

Provide centralized PSTN access

Centralize applications

Aggregate PBXs for migration to a UC system

All devices in this setup communicate using SIP. Avaya does not recognize the call-info header sent in the invite or the 200 ok causing basic audio or video calls to fail. As a work around, normalization script has been used on CISCO SME node to strip the CALL_INFO header leading to successful calls. Video calls between Avaya and Cisco are successful once we enable the video capable Codecs on endpoints (H.263 and H.264).

PSTN connectivity has been tested with Cisco UBE facing the service provider SIP trunk. The CUBE connects to SME and provides PSTN calling capabilities including fax from endpoints provisioned on Avaya and Cisco Unified Call Manager.

The following sections provide the required configurations for a successful integration.

Configuring the Avaya S8300 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.


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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.


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System Parameters IP Options


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IP Nodes


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IP Network Region Location:1

Authoritative Domain:lab.tekvizion.com

Name:tekvizion

Codec Set: 1

Inter/Intra-region IP-IP Direct Audio:YES


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IP Network Region (Contd …)

H.323 SECURITY PROFILES: any-auth


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IP Network Region (Contd …)


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IP Codec Set Codec set 1 is configured fior this test. Audio Codec G711MU and G.729 are select codec Media Encryption - None


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IP Codec set (Contd ...)

Allow Direct-IP Multimedia set to ‘y”

Set Maximum Call Rate for Direct-IP Multimedia:4096:Kbits

Set Maximum Call Rate for Priority Direct-IP Multimedia: 4096:Kbits

Note: The ip-codec-set configuration above is assigned to SIP trunk(s) using codecs G.729, 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”.


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Signaling Group

Set Group Type: sip IMS Enabled? N

Transport Method: tcp

IP Video? Y

Priority Video? Y

Peer Detection Enabled? Y

Near-end Node Name: procr

Far-end Node Name: SM1

Near-end Listen Port: 5060

Far-end Listen Port: 506

Far-end Network Region: 1

DTMF over IP: rtp-payload

Direct IP-IP Audio Connections? Y


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Trunk Group Group number: 4

Group Type:sip

Group Name:SIP to Cisco

TAC:*104

Member Assignment Method:auto

Service Type:tieSignaling Group:4

Number of Members:10


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Trunk Group (Contd …)

Preferred Minimum Session Refresh Interval(sec): 900


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Trunk Group (Contd …)

Numbering Format: private


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Trunk Group (Contd ...)

Mark Users as Phone? Y

Support Request History? Y

Telephone Event Payload Type: 101


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Route Pattern Pattern Number: 4

Pattern name: Cisco

Grp No: 4

FRL: 0

ITC:unre

Numbering Format:lev0-pvt

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


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Note: The Route Pattern configuration above is assigned to ARS analysis table entry 5, used to route calls to the SP using 1+10-digit dialing.


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AAR/ARS Analysis

AAR code 6 is used to route calls to Cisco UCM extensions over Route Pattern 4. This code will be used in the Uniform Dial plan table, and will be prefixed onto the 4-digit numbers assigned to Cisco UCM stations. 7000 used to reach the Voice Mail.


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Dial String 2142425 is used to route calls to the Service Provider. After dialing 9 (ARS access code) plus 21424259xx, 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 5.


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Uniform Dialing Plan


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ISDN Public/Unknown Numbering Plan 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 6XXX 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 4500 and 4501: when calls are sent over trunk group 4 (trunk group used to connect the PBX to Cisco UCM-SME).


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Incoming-call-handling-treatment 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 7193779211 , 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 9211 to ext. 4500.this is for as example we can translate the incoming number according to called numbers.


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Station Configuration (IP Phone)

Type : 630SIP

Port: S00121

Name: 4501, 4501

IP Soft Phone – Y

IP Video Soft Phone - Y


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Station Configuration (Analog Line - FAX) Type: 2500

Port: 001V301

Name: fax1


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Configuring the Avaya Aura – Session Manager Edition

Navigation: HomeElementsRouting


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Domains: Name: lab.tekvizion.com


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Locations: Name: Dallas Notes: Enterprise


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Adaptations: Adaptation for Cisco CUCM Adaptation name: Cisco-CUCM Module name: CiscoAdapter Module Parameter: fromto=true odstd=10.80.11.3 iosrcd=lab.tekvizion.com


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Adaptation for Avaya Aura CM Adaptation name:Avaya_CM Module name: DigitConversionAdapter Module Parameter: fromto=true


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SIP Entities: SIP Entity and entity Link for CUCM Name: Cisco_Cluster21A FQDN or IP Address: 10.80.11.2 Type: Other Adaptation: Cisco_CUCM10 Location: Dallas Time Zone: America/Chicago SIP Link Monitoring: Use Session Manager Configuration


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SIP Entity for Avaya Communications Manager Name: tekcm FQDN or IP Address: 10.70.2.14 Type: CM Adaptation: Avaya_CM Location: Dallas Time Zone: Chicago Sip Link Monitoring: Use Session Manager Configuration


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Entity Link between Avaya Session Manager and Cisco CUCM:

Name: teksm_clus21

SIP Entity 1: teksm

Protocol: tcp

Port: 5060

SIP Entity 2: CUCM10_CLUS21

Port: 5060

Trusted: checked


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Entity Link between Avaya Session manager and Avaya Aura Communication Manager:

Name:teksm_tekcm_5060_TCP

SIP Entity 1:teksm

Protocol:tcp

Port 5060

SIP Entity 2:tekcm

Port 5060

Trusted:checked


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Routing Policies Routing Policy for call to go to CUCM-SME

Name: to_CUCM10_CLUS21

Select SIP Entity “Cisco_Cluster21A” for SIP entity as destination


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Dial Pattern to reach CUCM: Pattern: 6 Min: 4 Max: 4 SIP Domain: lab.tekvizion.com Original Location Name: Dallas Routing Policy Name: to CUCM10_CLUS21


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Configuring the Cisco Unified Communications Manager – Session Manager Edition

1. Cisco UCM - Session Manager Edition software version 2. Partitions configuration 3. Calling Search Space configuration 4. Translation Pattern configuration 5. SIP profile (used by SIP trunks) configuration 6. SIP Normalization Script (used by SIP trunk to Avaya PBX) configuration 7. SIP trunk configuration to PSTN 8. SIP trunk configuration to Avaya PBX 9. SIP Trunk configuration to Cisco UCM 10. SIP Trunk configuration to Cisco Unity Connection 11. Route Pattern configuration to Avaya PBX 12. Route Pattern configuration to Cisco UCM 13. Route Pattern configuration to PSTN


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Cisco Unified Communications Manager – Session Manager Edition software version


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Configuration of Partitions Navigation Path: Call Routing Class of Control Partition

Partitions facilitate call routing by dividing the route plan into logical subsets that are based on organization, location, and call type


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Configuration of Calling Search Spaces Navigation Path: Call Routing Class of Control Calling Search Space A calling search space comprises an ordered list of route partitions that are typically assigned to devices. Calling search spaces determine the partitions that calling devices search when they are attempting to complete a call.

Below are two such calling search spaces created and Route Partitions have been assigned accordingly.


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Configuration of Calling Search Spaces (Contd...) Route Partition selected for calling search space = Translation Pattern – Route_Pattern


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Configuration of Calling Search Spaces (Contd...) Route Partition selected for calling search space = Trunks – Route_Pattern & Translation_Pattern


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Configuration of Translation Pattern used to strip leading digits on inbound calls from SP Navigation Path: Call Routing Translation Pattern 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. Set Translation Pattern = 719377.XXXX – This is defined to strip all leading digits for inbound calls from Service Provider network Description = Used to identify translation pattern


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Configuration of Translation Pattern used to strip leading digits on inbound calls from SP (Contd...) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation*= Allowed Discard digits = Predot – All the digits before the dot will be discarded


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Configuration of SIP Profile used by SIP trunks Navigation Path: Device Device Settings SIP Profile SIP Profile with Early offer enabled: Set Name* = Early Offer SIP Profile. This is used for this example Set Description = This is defined to identify this particular SIP Profile Disable Early Media on 180 – Check this box


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Configuration of SIP Profile used by SIP trunks (Contd…)

Set Accept Audio codec preferences in received offer* = Off. Should be turned on based up on requirement


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Configuration of SIP Profile used by SIP trunks (Contd…) Early offer Support for voice and video calls – Check this box to include SDP in INVITE for all outbound calls Enable OPTIONS Ping to monitor destination status for trunks with Service Type “None(Default)” – Check this box Send send-receive SDP in mid-call INVITE – Check this


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SIP Profile with Early Media disabled: Set Name*= SIP Profile with Early Media Disabled – This is used for this example Set Description – This is used to identify this particular SIP Profile Disable early Media on 180 – Check this box


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Configuration of SIP Profile used by SIP trunks (Contd…) Require SDP Inactive Exchange for Mid-call Media Exchange – Check this


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Set SIP Rel1xx Options = Send PRACK if 1xx contains SDP OPTIONS ping to monitor destination status for trunks – Check this


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Configuration of SIP Normalization Script (used by SIP trunk to Avaya PBX) Navigation Path: Device Device Settings SIP Normalization Script Normalization Script for CALL_INFO: Set Name* = CALL_INFO – This is used for this example Set Description = DELETE CALL_INFO – this is used to identify this particular script Set Content* = Please see the content in next page


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Complete Script: M = {} function M.outbound_INVITE(msg) msg:removeHeader("Call-Info") end function M.outbound_18X_INVITE(msg) msg:removeHeader("Call-Info") end function M.outbound_200_INVITE(msg) msg:removeHeader("Call-Info") end return M


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Configuration of SIP trunks to PSTN Navigation Path: Device Trunk Set Device Name* = Trunk_To_PSTN used for this example Set Desciption = This is used to identify the SIP trunk to PSTN Set Device Pool* = G711_pool ( should be set based up on requirement) Set Class Classification* = OffNet Set Media Resourece Group List = MRGL_MTP


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Configuration of SIP trunks to PSTN (Contd…) Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation* = Allowed


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Configuration of SIP trunks to PSTN (Contd…) Redirecting Diversion Header Delivery – Inbound = Check this check box to accept the Redirecting Number in the incoming SETUP message to the Cisco CallManager.


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Configuration of SIP trunks to PSTN (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Redirecting Diversion Header Delivery – Outbound = Check this check box to include the Redirecting Number IE in the outgoing SETUP message from the Cisco CallManager to indicate the first Redirecting Number and the redirecting reason of the call when the call is forwarded. Caller DN = 7193779211 – used for this example Set Destination Address = 10.64.3.43. Destination Port = 5060. This is used for this example Set SIP Trunk Security Profile* = Non Secure SIP Trunk Profile Set SIP Profile*= Early Offer SIP Profile Set DTMF Signaling Method*= RFC 2833


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Configuration of SIP trunks to PSTN (Contd…)


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Configuration of SIP trunk to Avaya PBX Navigation Path: Device Trunk Set Device Name* = Trunk_To_Avaya used for this example Set Desciption = This is used to identify the SIP trunk to Avaya Set Device Pool* = G711_pool ( should be set based up on requirement) Set Class Classification* = OnNet Set Media Resourece Group List = MRGL_MTP


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Configuration of SIP trunk to Avaya PBX (Contd…) Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation* = Allowed Redirecting Diversion Header Delivery – Inbound = Check this check box to accept the Redirecting Number in the incoming SETUP message to the Cisco CallManager.


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Configuration of SIP trunk to Avaya PBX (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Redirecting Diversion Header Delivery – Outbound = Check this check box to include the Redirecting Number IE in the outgoing SETUP message from the Cisco CallManager to indicate the first Redirecting Number and the redirecting reason of the call when the call is forwarded.


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Configuration of SIP trunk to Avaya PBX (Contd…) Set Destination Address = 10.70.2.6. This is used for this example Set SIP Trunk Security Profile* = Non Secure SIP Trunk Profile Set SIP Profile*= Early Offer SIP Profile Set DTMF Signaling Method*= RFC 2833


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Normalization Script = CALL_INFO

Configuration of SIP trunk to Cisco UCM Navigation Path: Device Trunk Set Device Name* = trunk to clus25 - used for this example Set Desciption = This is used to identify the SIP trunk to Avaya Set Device Pool* = G711_pool ( should be set based up on requirement) Set Class Classification* = OnNet


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Set Media Resourece Group List = MRGL_MTP


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Configuration of SIP trunk to Cisco UCM (Contd…) Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation* = Allowed Redirecting Diversion Header Delivery – Inbound = Check this check box to accept the Redirecting Number in the incoming SETUP message to the Cisco CallManager.


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Configuration of SIP trunk to Cisco UCM (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Redirecting Diversion Header Delivery – Outbound = Check this check box to include the Redirecting Number IE in the outgoing SETUP message from the Cisco CallManager to indicate the first Redirecting Number and the redirecting reason of the call when the call is forwarded.


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Configuration of SIP trunk to Cisco UCM (Contd…) Set Destination Address = 10.80.15.2. Destination Port = 5060 - This is used for this example Set SIP Trunk Security Profile* = Non Secure SIP Trunk Profile Set SIP Profile*= Early Offer SIP Profile Set DTMF Signaling Method* = RFC 2833


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Configuration of SIP trunk to Cisco UCM (Contd…)


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Configuration of SIP trunk to Unity Connection Navigation Path: Device -> Trunk


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Set Device Name* = Trunk_To_CUC- used for this example Set Desciption = This is used to identify the SIP trunk to Unity Connection Set Device Pool* = G729_pool ( should be set based up on requirement) Set Class Classification* = OnNet Set Media Resourece Group List = MRGL_MTP

Configuration of SIP trunk to Unity Connection (Contd…)


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Configuration of SIP trunk to Unity Connection (Contd…) Set Connected Line ID Presentation*= Allowed


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Set Connected Name Presentation* = Allowed Redirecting Diversion Header Delivery – Inbound = Check this check box to accept the Redirecting Number in the incoming SETUP message to the Cisco CallManager.



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Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Redirecting Diversion Header Delivery – Outbound = Check this check box to include the Redirecting Number IE in the outgoing SETUP message from the Cisco CallManager to indicate the first Redirecting Number and the redirecting reason of the call when the call is forwarded. Set Destination Address = 10.80.15.5 Destination port = 5060. This is used for this example Set SIP Trunk Security Profile = Non Secure SIP Trunk Profile Set SIP Profile= Early Offer SIP Profile DTMF signaling method = RFC2833



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Configuration of Route Patterns – To Avaya PBX


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Navigation Path: Call Routing Route/Hunt Route Pattern Set Route Pattern* = 4XXX This is used to route to Avaya in this example. Set Gateway/Route List* = Trunk_to_Avaya. This is used for this example. Class Classification* = OnNet Provide Outside Dial Tone – Uncheck this box


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Configuration of Route Patterns – To Avaya PBX (Contd…) Set Calling Line ID Presentation= Allowed Set Calling Name Presentation= Allowed Set Connected Line ID Presentation= Allowed Set Connected Name Presentation= Allowed


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Configuration of Route Patterns – To Cisco UCM Navigation Path: Call Routing Route/Hunt Route Pattern Set Route Pattern* =6XXX This is used to route to CUCM in this example. Set Gateway/Route List* = trunk_to_clus25. This is used for this example. Class Classification* = OnNet


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Configuration of Route Patterns – To Cisco UCM (Contd…) Provide Outside Dial Tone – Uncheck this box Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation*= Allowed


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Configuration of Route Patterns – To PSTN Navigation Path: Call Routing Route/Hunt Route Pattern Route Pattern to make National Calls: Set Route Pattern* = 1XXXXXXXXXX This is used to route to PSTN in this example. Set Gateway/Route List* = trunk_to_PSTN. This is used for this example. Class Classification* = OffNet


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Configuration of Route Patterns – To PSTN (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation*= Allowed


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Configuration of Route Patterns – To PSTN (Contd…) Route Pattern to make International calls: Set Route Pattern* = 011[2-9]! This is used to make international calls in this example Set Gateway/Route List* = trunk_to_PSTN. This is used for this example. Class Classification* = OffNet


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Configuration of Route Patterns – To PSTN (Contd…)

Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation*= Allowed


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Configuring the Cisco Unified Communications Manager

1. Cisco Unified Communications Manager Version 2. Service Parameters configuration 3. Audio Codec Preference List configuration 4. Region Configuration 5. Device Pool configuration 6. Conference Bridge configuration 7. Media Resource Group configuration 8. Media Resource Group List configuration 9. SIP Profile configuration 10. SIP Trunk to SME configuration 11. Route Pattern configuration to SP 12. Route Pattern configuration to Avaya 13. Cisco IP Phone 9971 SIP Configuration 14. Cisco IP Phone 8945 SCCP Configuration 15. MGCP Fax gateway configuration


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Cisco Unified Communications Manager Software Version


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Configuration of Service Parameters – Cisco Call Manager Navigation Path: System Service Parameters Set Duplex Streaming Enabled = True 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.


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Configuration of Audio Codec Preference List Navigation Path: System Region Information Audio Codec Preference List 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. G711_G729 and G729_G711 are the two Audio Codec Preferences configured


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Configuration of Audio Codec Preference List (Contd…) Set Name* = G711_G729. This is used for this example Set Description*= This text is used to identify this Audio codec preference list Set Codec in List = G.711 U-Law 64k. First choice in this example Set Codec in List = G.729 8k. Second choice in this example


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Configuration of Audio Codec Preference List (Contd…) Set Name = G729_G711. This is used for this example Set Description* = this text is used to identify this Audio codec preference list Set Codec in List = G.729 8k. First choice in this example Set Codec in List = G.711U-Law 64k. Second choice in this example


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Configuration of Regions Navigation Path: System Region Information Region G711_G729 and G729_G711 are the regions created in this example


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Configuration of Regions (Contd...) Set Name*= G711_G729. This is used in this example. Set Audio Codec Preference List= G711_G729 Set Maximum Audio Bit Rate= 64 Kbps (G7.22, G7.11). This is used in this example.


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Configuration of Regions (Contd...) Set Name*= G729_G711. This is used in this example. Set Audio Codec Preference List= G729_G711 Set Maximum Audio Bit Rate= 64 Kbps (G7.22, G7.11). This is used in this example.


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Configuration of Device Pool Navigation Path : System-> Device Pool G711_G729 and G729_G711 are the pools created in this example


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Configuration of Device Pool (Contd…) Set Device Pool Name*= G711_G729. This is used in this example. Set Cisco Unified Communications Manager Group*= Default Set Date/Time Group* = CMLocal Set Region* =G711_G729. This is used in this example Set Media Resource Group List = MRGL_MTP. This is used in this example.


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Configuration of Device Pool (Contd…)


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Configuration of Device Pool (Contd…) Set Device Pool Name*= G729_G711. This is used in this example. Set Cisco Unified Communications Manager Group*= Default Set Date/Time Group* = CMLocal Set Region* =G711_G729. This is used in this example Set Media Resource Group List = MRGL_MTP. This is used in this example.


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Configuration of Conference Bridge Navigation Path: Media Resources Conference Bridge Set Conference Bridge Name = CFB_2 Description = CFB_clus25pub Device Pool = G711_G729


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Configuration of Media Resource Group Navigation Path: Media Resources Media Resource Group Set Name*= MRG_RTP this is used for this example. Set Description = MRG with RTP - This text is used to define this Media Resource Group List. Set all Resources in the selected Media Resources Box


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Configuration of Media Resource Group List Navigation Path: Media Resources Media Resource Group List Set Name = MRGL_MTP Set selected Media Resource Groups – MRG_MTP


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Configuration of SIP Profile Navigation Path: Device Device Settings SIP Profile SIP Profile with Early Media Enabled: Set Name = Early Offer SIP Profile. This is used for this example Set Description = This text is used to identify SIP profile Disable Early Media on 180 – Check this box


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Configuration of SIP Profile (Contd…)


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Configuration of SIP Profile (Contd…) Early Offer support for voice and video calls (insert MTP if needed) – Check this box


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Configuration of SIP Profile (Contd…) Enable OPTIONS Ping to monitor destination status for Trunks with Service Type – Check this Send send-receive SDP in mid-call INVITE – Check this


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SIP Profile with Early Media Disabled: Set Name*= Standard SIP Profile w/early media disabled. This is used for this example. Set Description = Default SIP Profile w/early media disabled. This text is used to identify this SIP Profile. Disable Early Media on 180 = Check this box


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Configuration of SIP Profile (Contd…) Require SDP Inactive Exchange for Mid-Call Exchange = Check this box


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Configuration of SIP Profile (Contd…)


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Configuration of SIP Profile (Contd…) Enable OPTIONS Ping to monitor destination status for Trunks with Service Type "None (Default)" – Check this box


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Configuration of SIP Trunk Security Profile Set Name*= Non Secure SIP Trunk Profile. This is used for this example.

Set Description = This text is used to identify this SIP Trunk Security Profile.

Check Accept out of dialog refer

Check Accept unsolicited notification

Check Accept replaces header


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Configuration of SIP Trunk to SME Navigation Path: Device Trunk Set Device Name* = SIPTRUNK_To_SME used for this example Set Desciption = SIP Trunk to SME – This is used to identify the SIP trunk to PSTN Set Device Pool* = is set to G11_G729 ( should be set based up on requirement) Set Media Resourece Group List = MRGL_MTP


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Configuration of SIP Trunk to SME (Contd…) Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation* = Allowed


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Configuration of SIP Trunk to SME (Contd…) Redirecting Diversion Header Delivery – Inbound = Check this check box to accept the Redirecting Number in the incoming SETUP message to the Cisco CallManager.


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Configuration of SIP Trunk to SME (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Redirecting Diversion Header Delivery – Outbound = Check this check box to include the Redirecting Number IE in the outgoing SETUP message from the Cisco CallManager to indicate the first Redirecting Number and the redirecting reason of the call when the call is forwarded.


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Configuration of SIP Trunk to SME (Contd…) Set Destination Address = 10.80.11.2. This is used for this example Set SIP Trunk Security Profile* = UCM to SME SIP Trunk Profile Set SIP Profile*= Early Offer SIP Profile


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Configuration of Route Pattern to PSTN through SME Navigation Path: Call Routing Route/Hunt Route Pattern Set Route Pattern* = 9.@ This is used to route to PSTN via SME in this example. Set Numbering Plan = NANP used for this example Set Gateway/Route List* = SIPTRUNK_To_SME. This is used for this example. Class Classification = OffNet


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Configuration of Route Pattern to PSTN through SME (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation*= Allowed Set Discard Digits = Predot


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Configuration of Route Pattern to Avaya PBX through SME Navigation Path: Call Routing Route/Hunt Route Pattern Set Route Pattern* = 4xxx This is used to route to PSTN via SME in this example. Set Gateway/Route List* = SIPTRUNK_To_SME. This is used for this example. Class Classification* = OnNet


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Configuration of Route Pattern to Avaya PBX through SME (Contd…) Set Calling Line ID Presentation*= Allowed Set Calling Name Presentation*= Allowed Set Connected Line ID Presentation*= Allowed Set Connected Name Presentation*= Allowed


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Configuration of Cisco 9971 SIP Phone Navigation Path: Device Phone Set MAC Address* = C07BBCA1B811 . This is used in this example. Set Description = This text is used to identify this Phone


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Configuration of Cisco 9971 SIP Phone (Contd…) Set Device Pool*= G729_G711. This is used in this example. Set Phone Button Template*= Standard 9971 SIP. This is used in this example. Set Common Phone profile = Standard Common phone profile Set Media Resource Group List = MRGL_Default. This is used in this example. Set User Hold MOH Audio Source = 1-SampleAudioSource. Set Network Hold MOH Audio Source = 1-SampleAudioSource.


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Configuration of Cisco 9971 SIP Phone (Contd…) Uncheck Logged Into Hunt Group


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Configuration of Cisco 9971 SIP Phone (Contd…) Set Device Security Profile* = Cisco 9971- Standard SIP Non-Secure Profile. This is used in this example. Set SIP Profile*= Standard SIP Profile. This is used in this example.


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Configuration of Cisco 9971 SIP Phone (Contd…)


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Configuration of Cisco 9971 SIP Phone (Contd…) Set Cisco Camera* = Enabled. This is used in this example. Set Video Capablities* = Enabled. This is used in this example.


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Configuration of Cisco 9971 SIP Phone (Contd…) Set RTCP* = Enabled


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Configuration of Cisco 8945 SCCP Phone Navigation Path: Device Phone Set MAC Address* = 64E950CB2804 . This is used in this example. Set Description = This text is used to identify this Phone Set Device Pool*= G711_G729. This is used in this example.


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Configuration of Cisco 8945 SCCP Phone (Contd…) Set Phone Button Template*= Standard 8945 SCCP. This is used in this example. Set Common Phone profile = Standard Common phone profile Set Media Resource Group List = MRGL_Default. This is used in this example. Set User Hold MOH Audio Source = 1-SampleAudioSource. Set Network Hold MOH Audio Source = 1-SampleAudioSource.


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Configuration of Cisco 8945 SCCP Phone (Contd…) Uncheck Logged on to Hunt Group


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Configuration of Cisco 8945 SCCP Phone (Contd…)


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Configuration of Cisco Unity Connection User Set Alias*= 6002. This is used for this example.

Set First Name = This text is used to identify this User.

Set Last Name* = cisco This is used for this example

Set Display Name= Test3 cisco. This is used in this example. Set SMTP Address =6002. This is used in this example. Set Phone System= CUCM. This is used in this example.


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Configuration of Cisco Unity Connection User (Contd ..) Set Extension = 6002 Set Phone system = CUCM


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Configuration of Cisco Unity Connection User(6002) (Contd ..)


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Configuration of FAX Gateway

Navigation Path: Device Gateway Set Domain Name = FAX-Gateway.lab.tekvizion.com = Used for this example Description – this is used to identify FAX Gateway Subunit 0 = VIC3-2FXS/DID


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Configuration of FAX Gateway (Contd…) Set Modem Passthrough = Enable Set Cisco Fax Relay = Disable Set T38 Fax Relay = Enable


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Configuration of FAX Gateway Analog Endpoint Navigation Path: Device Gateway Set End-Point Name = AALN/S0/SU0/[email protected] = Used for this example Set Description = this is used to identify end point type Set Media Resource Group List = MRGL_MTP

mailto:AALN/S0/SU0/[email protected]


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Configuration of FAX Gateway Analog Endpoint (Contd…)


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FAX Gateway Configuration: IF YOU DO NOT CHANGE THE PUBLICLY-KNOWN CREDENTIALS, YOU WILL NOT BE ABLE TO LOG INTO THE DEVICE AGAIN AFTER YOU HAVE LOGGED OFF. For more information about Cisco CP please follow the instructions in the QUICK START GUIDE for your router or go to http://www.cisco.com/go/ciscocp ----------------------------------------------------------------------- User Access Verification Username: cisco Password: % Password expiration warning. ----------------------------------------------------------------------- Cisco Configuration Professional (Cisco CP) is installed on this device and it provides the default username "cisco" for one-time use. If you have already used the username "cisco" to login to the router and your IOS image supports the "one-time" user option, then this username has already expired. You will not be able to login to the router with this username after you exit this session. It is strongly suggested that you create a new username with a privilege level of 15 using the following command. username <myuser> privilege 15 secret 0 <mypassword> Replace <myuser> and <mypassword> with the username and password you want to use. -----------------------------------------------------------------------

FAX-GATEWAY# FAX-GATEWAY# FAX-GATEWAY#show mgcp MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE MGCP call-agent: clus25sub1 2427 Initial protocol service is MGCP 0.1 MGCP validate call-agent source-ipaddr DISABLED MGCP validate domain name DISABLED MGCP block-newcalls DISABLED MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED MGCP quarantine mode discard/step MGCP quarantine of persistent events is ENABLED


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MGCP dtmf-relay voip codec all mode out-of-band MGCP dtmf-relay for voAAL2 is SDP controlled MGCP voip modem passthrough disabled MGCP voaal2 modem passthrough disabled MGCP voip modem relay: Disabled MGCP T.38 Named Signalling Event (NSE) response timer: 200 MGCP Network (IP/AAL2) Continuity Test timer: 200 MGCP 'RTP stream loss' timer disabled MGCP request timeout 500 MGCP maximum exponential request timeout 4000 MGCP rtp unreachable timeout 1000 action notify MGCP gateway port: 2427, MGCP maximum waiting delay 3000 MGCP restart delay 0, MGCP vad DISABLED MGCP rtrcac DISABLED MGCP system resource check DISABLED MGCP xpc-codec: DISABLED, MGCP persistent hookflash: DISABLED MGCP persistent offhook: ENABLED, MGCP persistent onhook: DISABLED MGCP piggyback msg ENABLED, MGCP endpoint offset DISABLED MGCP simple-sdp ENABLED MGCP undotted-notation DISABLED MGCP codec type g711ulaw, MGCP packetization period 20 MGCP JB threshold lwm 30, MGCP JB threshold hwm 150 MGCP LAT threshold lwm 150, MGCP LAT threshold hwm 300 MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000 MGCP CL threshold lwm 1000, MGCP CL threshold hwm 10000 MGCP playout mode is adaptive 60, 40, 1000 in msec MGCP Fax Playout Buffer is 300 in msec MGCP media (RTP) dscp: ef, MGCP signaling dscp: af31 MGCP default package: fxr-package MGCP supported packages: gm-package dtmf-package trunk-package line-package hs-package rtp-package atm-package ms-package dt-packag e mo-package mt-package sst-package fxr-package pre-packa ge md-package MGCP Digit Map matching order: shortest match SGCP Digit Map matching order: always left-to-right MGCP VoAAL2 ignore-lco-codec DISABLED MGCP T.38 Max Fax Rate is DEFAULT MGCP T.38 Fax is ENABLED MGCP T.38 Fax ECM is DISABLED MGCP T.38 Fax NSF Override is ENABLED: 000000 MGCP T.38 Fax Low Speed Redundancy: 2 MGCP T.38 Fax High Speed Redundancy: 2 MGCP Fax relay SG3-to-G3: ENABLED


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MGCP Fax relay ANSam suppression: DISABLED MGCP control bound to interface GigabitEthernet0/0 MGCP media bound to interface GigabitEthernet0/0 MGCP Upspeed payload type for G711ulaw: 0, G711alaw: 8 MGCP Dynamic payload type for Cisco fax indication: 96, Cisco fax ack: 97 MGCP Dynamic payload type for G.726-16K codec MGCP Dynamic payload type for G.726-24K codec MGCP Dynamic payload type for G.726-32K codec MGCP Dynamic payload type for G.Clear codec MGCP Dynamic payload type for NSE is 100 MGCP Dynamic payload type for NTE is 101 MGCP rsip-range is enabled for TGCP only. MGCP Comedia role is NONE MGCP Comedia check media source is DISABLED MGCP Comedia SDP force is DISABLED MGCP g729 variants use static payload type and contain annexb in SDP MGCP Guaranteed scheduler time is DISABLED MGCP Disconnect delay error recovery DISABLED MGCP DNS stale threshold is 30 seconds


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FAX-GATEWAY#sh run Building configuration... Current configuration : 6614 bytes ! ! Last configuration change at 04:48:08 CST Wed Feb 12 2014 by cisco version 15.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname FAX-GATEWAY ! boot-start-marker boot-end-marker ! ! logging buffered 51200 warnings enable secret 4 iR3uUX3Bo6oYbT6ajhFwJe39FR4g.1QCmm7yYduKGZI ! no aaa new-model clock timezone CST -6 0 clock summer-time CDT recurring ! ip cef ! ! ! ! ! ! ip domain name lab.tekvizion.com ip name-server 10.64.1.3 no ipv6 cef multilink bundle-name authenticated ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-1120430079 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-1120430079


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revocation-check none rsakeypair TP-self-signed-1120430079 ! ! crypto pki certificate chain TP-self-signed-1120430079 certificate self-signed 01 3082022B 30820194 A0030201 02020101 300D0609 2A864886 F70D0101 05050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 31313230 34333030 3739301E 170D3133 31303031 32313434 33305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 31323034 33303037 3930819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100D201 333D5645 360F1451 B7BC6EBB 7FFC6715 F1658731 5ABE4743 6B6EA3B0 1C7D1D76 17B5BD08 BA68F94B 80F790EA 8C46036E 7FDE5827 3BF9B45F 7B71D959 B6943C33 A4781184 7DA425F8 81EC15A8 1AFC1E03 3690A9E4 DF1C9AF1 974151B1 15742950 E0ABEBBB DDE57C06 97D3C3A4 4205ADE2 36CB165E 47967DC4 8F41F25B 17430203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF 301F0603 551D2304 18301680 14107F60 ACB1F013 843D556C F6C614CF 491E69DA E5301D06 03551D0E 04160414 107F60AC B1F01384 3D556CF6 C614CF49 1E69DAE5 300D0609 2A864886 F70D0101 05050003 81810029 2333B82C 9C91341D 6921E674 E365F73A 840DB5A3 54C20F8B E004BE5E 741B5B53 3978D629 DBE53B11 51C2F563 9A82DCC4 37D269C6 B8A5F80D 969C90E1 DA963C31 D49B3F9C B7A1E484 C792A0E7 1BEA5D90 0A501640 BD4045A8 649E269C B4891C0D FF317730 DD446398 2E0130DB 6DCD39EA 8C68EC80 308E03E1 80C6C503 CD45A4 quit voice-card 0 ! ! ! voice service voip no supplementary-service sip handle-replaces fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw no fax-relay sg3-to-g3 ! ! ! ! ! ! license udi pid CISCO2901/K9 sn FTX174081TS hw-module pvdm 0/0 ! ! ! username cisco privilege 15 secret 4 tnhtc92DXBhelxjYk8LWJrPV36S2i4ntXrpb4RFmfqY


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! redundancy ! ! ! ! ! ! interface Embedded-Service-Engine0/0 no ip address shutdown ! interface GigabitEthernet0/0 description FAX Gateway for UCM-SME ip address 10.80.15.7 255.255.255.0 duplex auto speed auto ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! 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 route 10.64.0.0 255.255.0.0 10.80.15.1 ip route 10.70.2.0 255.255.255.0 10.80.15.1 ip route 10.80.11.0 255.255.255.0 10.80.15.1 ip route 172.16.31.0 255.255.255.0 10.80.15.1 ! ! ! ! control-plane ! ! voice-port 0/0/0 timing hookflash-in 1550 150


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station-id name fax test ! voice-port 0/0/1 ! ! ! ! ccm-manager fallback-mgcp ccm-manager redundant-host clus25sub2 clus25pub ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server clus25pub.lab.tekvizion.com ccm-manager config ! mgcp mgcp call-agent clus25pub 2427 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp rtp unreachable timeout 1000 action notify mgcp modem passthrough voip mode nse mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package mgcp default-package fxr-package no mgcp package-capability res-package no mgcp timer receive-rtcp mgcp sdp simple mgcp fax rate 9600 mgcp fax t38 ls_redundancy 2 mgcp fax t38 hs_redundancy 2 mgcp fax t38 nsf 000000 mgcp fax t38 gateway force mgcp bind control source-interface GigabitEthernet0/0 mgcp bind media source-interface GigabitEthernet0/0 mgcp rtp payload-type nte 101 ! mgcp profile default ! ! dial-peer voice 999000 pots service mgcpapp destination-pattern 6100 port 0/0/0 ! !


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! ! gatekeeper shutdown ! ! banner exec ^C % Password expiration warning. ----------------------------------------------------------------------- Cisco Configuration Professional (Cisco CP) is installed on this device and it provides the default username "cisco" for one-time use. If you have already used the username "cisco" to login to the router and your IOS image supports the "one-time" user option, then this username has already expired. You will not be able to login to the router with this username after you exit this session. It is strongly suggested that you create a new username with a privilege level of 15 using the following command. username <myuser> privilege 15 secret 0 <mypassword> Replace <myuser> and <mypassword> with the username and password you want to use. ----------------------------------------------------------------------- ^C banner login ^C ----------------------------------------------------------------------- Cisco Configuration Professional (Cisco CP) is installed on this device. This feature requires the one-time use of the username "cisco" with the password "cisco". These default credentials have a privilege level of 15. YOU MUST USE CISCO CP or the CISCO IOS CLI TO CHANGE THESE PUBLICLY-KNOWN CREDENTIALS Here are the Cisco IOS commands. username <myuser> privilege 15 secret 0 <mypassword> no username cisco Replace <myuser> and <mypassword> with the username and password you want to use. IF YOU DO NOT CHANGE THE PUBLICLY-KNOWN CREDENTIALS, YOU WILL NOT BE ABLE


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TO LOG INTO THE DEVICE AGAIN AFTER YOU HAVE LOGGED OFF. For more information about Cisco CP please follow the instructions in the QUICK START GUIDE for your router or go to http://www.cisco.com/go/ciscocp ----------------------------------------------------------------------- ^C ! line con 0 exec-timeout 0 0 login local line aux 0 line 2 no activation-character no exec transport preferred none transport output pad telnet rlogin lapb-ta mop udptn v120 ssh stopbits 1 line vty 0 4 login local transport input telnet ssh line vty 5 15 privilege level 15 login local transport input telnet ssh ! scheduler allocate 20000 1000 ! end FAX-GATEWAY# FAX-GATEWAY#


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Configuring the Cisco UBE - Enterprise

User Access Verification

Username: cisco

Password:

CUCM-SME-Avaya#show version

Cisco IOS Software, C2951 Software (C2951-UNIVERSALK9-M), Version 15.3(3)M, RELEASE SOFTWARE (fc2)

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

Copyright (c) 1986-2013 by Cisco Systems, Inc.

Compiled Mon 22-Jul-13 04:58 by prod_rel_team

ROM: System Bootstrap, Version 15.0(1r)M6, RELEASE SOFTWARE (fc1)

CUCM-SME-Avaya uptime is 4 weeks, 5 days, 22 hours, 1 minute

System returned to ROM by reload at 12:08:44 UTC Thu Dec 26 2013

System image file is "flash0:c2951-universalk9-mz.SPA.153-3.M.bin"

Last reload type: Normal Reload

Last reload reason: Reload Command

This product contains cryptographic features and is subject to United

States and local country laws governing import, export, transfer and

use. Delivery of Cisco cryptographic products does not imply

third-party authority to import, export, distribute or use encryption.

Importers, exporters, distributors and users are responsible for

compliance with U.S. and local country laws. By using this product you


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agree to comply with applicable laws and regulations. If you are unable

to comply with U.S. and local laws, return this product immediately.

A summary of U.S. laws governing Cisco cryptographic products may be found at:

http://www.cisco.com/wwl/export/crypto/tool/stqrg.html

If you require further assistance please contact us by sending email to

[email protected].

Cisco CISCO2951/K9 (revision 1.1) with 1835264K/261888K bytes of memory.

Processor board ID FTX1509AJJV

3 Gigabit Ethernet interfaces

1 terminal line

DRAM configuration is 72 bits wide with parity enabled.

255K bytes of non-volatile configuration memory.

250880K bytes of ATA System CompactFlash 0 (Read/Write)

License Info:

License UDI:

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

Device# PID SN

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

*0 CISCO2951/K9 FTX1509AJJV


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Technology Package License Information for Module:'c2951'

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

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

appx None None None

NtwkEss None None None

CollabPro None None None

Configuration register is 0x2102

User Access Verification Username: cisco Password: CUCM-SME-Avaya#sh run Building configuration... Current configuration : 5413 bytes ! ! Last configuration change at 09:08:13 UTC Wed Feb 12 2014 by cisco version 15.3 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname CUCM-SME-Avaya !


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boot-start-marker boot-end-marker ! aqm-register-fnf ! enable secret 4 sKPgCY/XPea3wk8xoeSWo7UGFaNVwzXDEyXWhuDjeLk ! no aaa new-model ! ! ! ! ! ! ! ip domain name lab.tekvizion.com ip name-server 10.64.1.3 ip cef no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! ! password encryption aes ! ! voice-card 0 dsp services dspfarm ! ! ! voice service voip ip address trusted list ipv4 0.0.0.0 no ip address trusted authenticate mode border-element license capacity 200 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback none sip


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session refresh registrar server expires max 1200 min 300 midcall-signaling passthru ! voice class codec 1 codec preference 1 g729r8 bytes 30 codec preference 2 g711ulaw ! voice class codec 2 codec preference 1 g711ulaw codec preference 2 g729r8 bytes 30 ! ! ! ! ! ! license udi pid CISCO2951/K9 sn FTX1509AJJV hw-module pvdm 0/0 ! ! ! username CISCO privilege 15 password 0 tekV1z10n ! redundancy ! ! ! translation-rule 719377921 Rule 1 7193779211 4100 Rule 2 7193779212 6100 Rule 3 7193779213 6002 ! ! ! ! ! interface Embedded-Service-Engine0/0 no ip address shutdown ! interface GigabitEthernet0/0 description WAN Interface to Verizon Trunk ip address 97.79.185.172 255.255.255.128 duplex auto


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speed auto ! interface GigabitEthernet0/1 description LAN Interface to CiscoSME Network ip address 10.64.3.43 255.255.0.0 duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/2 no ip address shutdown duplex auto speed auto ! ip default-gateway 97.79.185.129 ip forward-protocol nd ! no ip http server no ip http secure-server ! ip route 0.0.0.0 0.0.0.0 97.79.185.129 ip route 10.80.11.0 255.255.255.0 10.64.1.1 ip route 10.85.0.0 255.255.0.0 10.64.1.1 ip route 172.16.31.0 255.255.255.0 10.64.1.1 ! ! nls resp-timeout 1 cpd cr-id 1 ! access-list 1 permit any ! control-plane ! ! ! ! ! ! ! mgcp behavior rsip-range tgcp-only mgcp behavior comedia-role none mgcp behavior comedia-check-media-src disable mgcp behavior comedia-sdp-force disable !


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mgcp profile default ! sccp local GigabitEthernet0/1 sccp ccm 10.80.11.3 identifier 1 priority 1 version 7.0 ! sccp ccm group 40 description SME MTP bind interface GigabitEthernet0/1 associate ccm 1 priority 1 associate profile 40 register SMEMTP ! dspfarm profile 40 mtp codec pass-through codec g711ulaw maximum sessions software 20 associate application SCCP shutdown ! dial-peer voice 214242596 voip description CUCM-SME-Avaya-Veri destination-pattern 21424259[0-9][0-9] session protocol sipv2 session target ipv4:63.87.147.62:5071 session transport udp voice-class codec 2 no voice-class sip early-offer forced dtmf-relay rtp-nte fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback none no vad ! dial-peer voice 10001 voip description CUCM-SME-Avaya-Veri-international destination-pattern 011............ session protocol sipv2 session target ipv4:63.87.147.62:5071 session transport udp voice-class codec 2 voice-class sip early-offer forced dtmf-relay rtp-nte ! dial-peer voice 10002 voip description CUCM-SME-Avaya-Veri-international destination-pattern 800....... session protocol sipv2 session target ipv4:63.87.147.62:5071


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session transport udp voice-class codec 2 voice-class sip early-offer forced dtmf-relay rtp-nte ! dial-peer voice 7323204 voip description CUCM-SME-AT&TIPTC destination-pattern 7323204[0-9][0-9][0-9] session protocol sipv2 session target ipv4:63.87.147.62:5071 session transport udp voice-class codec 2 voice-class sip early-offer forced dtmf-relay rtp-nte ! dial-peer voice 719377921 voip description CUBE-SME-CUCM/Avaya destination-pattern 719377921[1-3] translate-outgoing called 719377921 session protocol sipv2 session target ipv4:10.80.11.2 voice-class codec 1 voice-class sip early-offer forced dtmf-relay rtp-nte fax rate 14400 fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback none no vad ! dial-peer voice 9729 voip description CUCM-SME-Avaya-Veri destination-pattern 9729655928 session protocol sipv2 session target ipv4:63.87.147.62:5071 session transport udp voice-class codec 2 no voice-class sip early-offer forced dtmf-relay rtp-nte ! dial-peer voice 10004 voip ! dial-peer voice 104 voip description CUCM-SME-Avaya-Veri destination-pattern 1.......... session protocol sipv2 session target ipv4:63.87.147.62:5071


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session transport udp voice-class codec 2 voice-class sip early-offer forced dtmf-relay rtp-nte ! dial-peer voice 105 voip description CUCM-SME-Avaya-Veri destination-pattern 0 session protocol sipv2 session target ipv4:63.87.147.62:5071 session transport udp voice-class codec 2 voice-class sip early-offer forced dtmf-relay rtp-nte ! ! sip-ua ! ! ! gatekeeper shutdown ! ! ! line con 0 line aux 0 line 2 no activation-character no exec transport preferred none transport output pad telnet rlogin lapb-ta mop udptn v120 ssh stopbits 1 line vty 0 4 login local transport input all ! scheduler allocate 20000 1000 ! end CUCM-SME-Avaya#


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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


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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.

Documents

Avaya S8300 Release 6.0.1 using SIP via Cisco Unified ... · © 2014 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems,