AT&T IP FlexReach: Connecting Cisco Unified Communications Manager 7.1 via the Cisco Unified Border Element 1.3 using SIP [Products & Services]

AT&T IP FlexReach: Connecting Cisco Unified Communications Manager 7.1 via the Cisco Unified Border Element 1.3 using SIP

July 17, 2009

Table of Contents

Introduction

Service Providers today, such as AT&T, are offering alternative methods to connect to the PSTN via their IP network. Most of these services utilize SIP as the primary signaling method and a centralized IP to TDM gateway to provide on-net and off-net services. AT&T IP FlexReach is a service provider offering that allows connection to the PSTN and may offer the end customer a viable alternative to traditional PSTN connectivity via either analog or T1 lines. A demarcation device between these services and customer owned services is recommended. The Cisco Unified Border Element provides demarcation, security, interworking and session management services.

· This application note describes how to configure a Cisco Unified Communications Manager (CUCM) 7.1 with a Cisco Unified Border Element (CUBE) 1.3 for connectivity to AT&T's IP Flex-Reach SIP trunk service. The application note also covers support and configuration examples of Cisco Unified MeetingPlace 7.0 and Cisco Unity messaging integrated to the Cisco Unified Communications Manager. The deployment model covered in this application note is Customer Premises Equipment (Unity/Meetingplace/CUCM/CUBE) to PSTN (AT&T IP Flex-Reach SIP). AT&T IP Flex-Reach provides inbound and outbound call service. This document does not address 911 emergency outbound calls. For 911 feature service details contact AT&T directly.

Note: The results and configuration samples are based on CUCM 7.0 and CUBE 1.2 releases but was determined by Cisco that changes done on the CUCM 7.1 and CUBE 1.3 releases will not affect the interoperability of SIP trunking to AT&T IP Flex-Reach services.

· Testing was performed in accordance to AT&T's IP Flex-Reach test plan and all features were verified. Key features verified are: inbound and outbound basic call (including international calls), calling name delivery, calling number and name restriction, DNIS translations, CODEC negotiation, advanced 8YY call prompter, intra-site transfers, intra-site conferencing, call hold and resume, call forward (forward all, busy and no answer), CPE offered scheduled and reservationless audio conferencing (Cisco MeetingPlace), leaving and retrieving voicemail (Cisco Unity), auto-attendant (Cisco Unity), fax using T.38 and G.711 (G3 and SG3 speeds), teleconferencing and outbound/inbound calls to/from TDM networks.

· The Cisco Unified Border Element configuration detailed in this document is based on a lab environment with a simple dial-plan used to ensure proper interoperability between AT&T SIP network and Cisco Unified Communications. The configuration described in this document details the important commands for successful interoperability, and care must be taken by the network administrator deploying CUBE to ensure these commands are set per each dial-peer requiring to interoperate to AT&T SIP network.

· This application note does not cover the use of calling search spaces (CSS) or partitions on Cisco Unified Communications Manager. To understand and learn how to apply CSS and partitions refer to the cisco.com link below.

Cisco Unified Communications Manager Administration Guide, Release 7.1(2) [Cisco Unified Communications Manager (CallManager)] (http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/bccm-712-cm.html)

Network Topology

Figure 1. Basic Call Setup

System Components

Hardware Components

· Cisco IOS gateway running CUBE 1.3 (IOS _ivs image version 12.4(22)YB4 or later)

· Cisco Unified Border Element is an integrated Cisco IOS Software application that runs on various IOS platforms, follow the link for more details: http://www.cisco.com/go/cube

· Packet Voice Data Module (PVDM). You will need to install DSP modules (PVDM) on CUBE 1.3 if you require MTP, Transcoding or Conference Bridge resources for codecs other than G.711. DSP are not required for basic calls. Follow the link for system required DSP calculator. http://www.cisco.com/pcgi-bin/Support/DSP/cisco_dsp_calc.pl

· Cisco MCS 7800 Series server (Cisco Unified Communications Manager)

· Cisco MCS 7800 Series server (Cisco Unified MeetingPlace Application Server)

· Cisco Unified MeetingPlace Media Server 3545 with (1) Audio Blade

· Cisco MCS 7800 Series Server (Cisco Unity)

· Cisco IP Phones ( The topology diagram shows 7960 and 7961, but any Cisco IP phone model supporting RFC2833 can be used)

· Cisco IOS Gateway (only needed if fax, analog phones or TDM systems are to interconnect). This component may be a H323, SIP or MGCP gateway, the protocol is optional and the choice is left up to the customer's network design.

Software Requirements

· Cisco Unified CM 7.1 and later 7.x releases. This solution was tested with 7.0.1.21007-1

· CUBE version 1.3 IOS version 12.4(22)YB4 or later. This configuration was tested with ADVANCED ENTERPRISE SERVICES (adventerprisek9_ivs-mz.)

· Cisco IOS gateway (IP-TDM) version: 12.4 or later.

· The documented CUBE configuration can be supported with the following IOS feature sets: IP VOICE, SP SERVICES, ADVANCED IP SERVICES, ADVANCED ENTERPRISE SERVICES, INT VOICE/VIDEO, IPIP GW, TDMIP GW,INT VOICE/VIDEO, IPIPGW, TDMIP GW AES

· Cisco Unified MeetingPlace 7.0.1 Applications Server software and Media Server Software

· Cisco Unity 7.0 software

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

Features

Features Supported

· Basic Call using G.729

· Calling Party Number Presentation and Restriction

· Calling Name

· AT&T Advanced 8YY Call Prompter (8YY)

· Intra-site Call Transfer

· Intra-site Conference, see caveat section for details.

· Call Hold and Resume

· Call Forward All, Busy and No Answer

· AT&T IP Teleconferencing

· CPE offered Scheduled/Reservationless conferencing (Cisco Unified Meetingplace)

· Fax using T.38 or G.711

· Incoming DNIS Translation and Routing

· CUBE: performs Delayed-Offer-to-Early-Offer conversion of an initial SIP INIVTE without SDP

· Outbound calls to AT&T's IP and TDM networks

· CPE voicemail managed service, leave and retrieve voice messages via incoming AT&T SIP trunk (Cisco Unity)

· Auto-attendant transfer-to service (Cisco Unity)

Features Not Supported

· CUCM/CUBE Codec negotiation of G.726

Caveats

· When using G.729 between AT&T IP Flex-Reach and Cisco Unified Border Element/Cisco Unified Communications Manager SIP trunk it is required to configure a conference bridge (CFB) resource on CUBE in order for Cisco Unified Communications Manager IP phone to initiate a three-way conference between G729 media end-points. See configuration section for details.

· Cisco Unified IP phones using SIP as the registration protocol (SIP-line) do not support G.729 with annex B. This current SIP line side support causes failed call attempts when CUBE is set for codec "g729br8" negotiation. Workaround is to remove "g729br8" from the preference codec list and only enable "g729r8". See configuration section for details.

· For DTMF digit passing using RFC2833 you must set a payload-type value of 100 for "nte" (named-telephone-events DTMF) on CUBE dial-peer pointing towards Cisco Unified Communications Manager. AT&T has performed a best-effort attempt to harmonize it's IP Flex-Reach SIP network to utilize Payload-type value (PT value)100 for RFC2833 (DTMF) application. If you find you have an issue with DTMF tones after you have configured the dial-peer for PT value=100, as per the configuration section, please see Appendix A for a step-by-step guide on how to troubleshoot and obtain the PT value being negotiated within your "live" network and how to change your dial-peer settings to the newly acquired PT value.

· In an incoming call to Meetingplace from AT&T SIP trunk (IP Flex-reach) the Meetingplace server will force a payload-type value "101" for NTE events (RFC2833) causing DTMF interworking failure. The workaround is to create a dedicated dial-peer, on CUBE, for incoming calls towards Meetingplace and force payload-type value "101" towards the AT&T SIP trunk (IP Flex-reach service) causing the AT&T SIP trunk service to utilize payload-type value 101 instead of 100 for the established call with Meetingplace. See configuration section for details.

· For forwarded calls from CUCM user to PSTN (out to AT&T's IP Flex-reach service) some AT&T serviced areas require that the SIP Diversion header contain the full 10-digit DID number of the forwarding party. In this application note the assumption was made that a typical customer will utilize extension numbers (4-digit assignments in this example) and map 10-digit DID number using CUCM translation pattern. Because we use 4-digit extensions on our CUCM IP phones it is necessary to expand the 4-digit extension, included in the Diversion: header of a forwarding INVITE message, to its full 10-digit DID number when the IP phone is set to call-forward. The requirement to expand the Diversion-Header has been achieved by the use of a SIP profile in CUBE (See configuraion section for details.). Alternatives methods, such as the recommended use of a translation profile (see Appendix A for a sample configuration) have been inconsistent (CSCsx62600) and as such using a SIP profile is the recommended solution.

· The CUBE depicted in figure 1 is not an AT&T managed device. It is recommended that the group responsible for the administration, management and configuration of the Cisco Unified Communications Manager, also manage and configure the Cisco Unified Border Element.

· While AT&T IP Flexible Reach services support E911/911 calling capabilities in certain circumstances, there are significant limitations on how these capabilities are delivered. Please review the AT&T IP Flexible Reach Service Guide in detail to understand these limitations and restrictions.

Configuration

Cisco IOS version

c3825_CUBE#sh ver

Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS-M), Version 12.4(22

)T, RELEASE SOFTWARE (fc1)

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

Compiled Fri 10-Oct-08 06:43 by prod_rel_team

ROM: System Bootstrap, Version 12.3(11r)T2, RELEASE SOFTWARE (fc1)

c3825_CUBE uptime is 2 weeks, 1 day, 3 hours, 37 minutes

System returned to ROM by power-on

System image file is "flash:c3825-adventerprisek9_ivs-mz.124-22.YB.bin"

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

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

export@cisco.com.

Cisco 3825 (revision 1.1) with 487423K/36864K bytes of memory.

Processor board ID FTX1025A25D

2 Gigabit Ethernet interfaces

1 Virtual Private Network (VPN) Module

DRAM configuration is 64 bits wide with parity enabled.

479K bytes of NVRAM.

62720K bytes of ATA System CompactFlash (Read/Write)

Configuration register is 0x2102

Configuring Cisco Unified Border Element (CUBE)

Critical commands are marked bold with footnote and description at bottom of the page

c3825_CUBE#sh run

Building configuration...

Current configuration : 4939 bytes

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

hostname c3825_CUBE

boot-start-marker

boot-end-marker

logging message-counter syslog

logging buffered 100000000

no aaa new-model

dot11 syslog

ip source-route

ip cef

no ipv6 cef

multilink bundle-name authenticated

voice-card 0

dspfarm

dsp services dspfarm¹

voice service voip

address-hiding²

allow-connections sip to sip³

redirect ip2ip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw⁴ è

h323

sip

header-passing error-passthru⁵

asserted-id pai⁶

midcall-signaling passthru⁷

privacy-policy passthru⁸

g729 annexb-all⁹

voice class codec 1¹⁰

codec preference 1 g729r8

codec preference 2 g711ulaw

voice class sip-profiles 1

INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:732320\1@\2>¹¹"

Configuring the Cisco Unified Communications Manager

Cisco Unified Communications Manager version

Version 7.1(2), 7.1(2a)

Single g729 SIP trunk configuration (Title page)

Single g729 SIP Trunk to AT&T (CUBE) configuration

Note: For caller id to be delivered using P-asserted-id (PAI) method, set values as per the above page and configure CUBE appropriately for PAI interworking. To enable caller id restriction using PAI privacy setting you must enable the privacy setting at the CUBE. If privacy is set at this CUCM configuration page, the privacy setting will not be sent to the AT&T SIP trunk side and the caller id will be presented. Please see caveat section and CUBE configuration for details.

Note: Ensure destination IP address is CUBE's private side IP address to where outgoing calls should be routed. Notice device pool is set to CUBE where region setting is CUBE (g729).

Configuring two SIP trunks to enable support for G729 and G711 end-points

Note: For these particular SIP trunk configurations and applications, the CUCM on-board MTP resource must not be part of the device pool for either Sip trunk configured in these steps. In this example the g711 SIP trunk (c3825_CUBE_g711) utilizes the "Default" device pool for g711 codec region. Because the on-board MTP resource is part of the "Default" device pool by default a bogus media resource group was created and the CUCM on-board MTP resource was moved to this bogus media resource group, in order to prevent undesired functionality of codec negotiation.G729 SIP trunk

Note: Device Pool "CUBE" enables G729 codec on this trunk

Note: Ensure destination IP address is CUBE's private side IP address to where outgoing calls should be routed

G711 SIP trunk

Note: Device Pool "Default" enables G711 codec on this trunk

Note: Ensure destination IP address is CUBE's private side IP address to where outgoing calls should be routed Configuring Route Group and Route List linking G729 and G711 SIP trunks to a single outbound trunk

Route Group

Note: The g729 and g711 SIP trunks created in the previous steps shall be assigned to the "Selected Devices" section of this configuration page.

Route List

Note: Name assigned to the Route List will be the device name you will need to assign on all route patterns for outgoing calls (calls to AT&T SIP service).

Configuring SIP trunk to Cisco Unified Meetingplace

Note: Ensure destination IP address is the address of the MeetingPlace Application server where incoming Meetingplace calls should be routed to. Notice the device pool is set to CUBE where region setting is CUBE (g729).

Region main page (codec settings)

Note: During testing three "device pools" were created. Within the device pool settings a "region" matching the device pool name was assigned to each device pool. Codec settings were assigned to devices using the region assignment and the relationship of each region to other regions. See further down for configuration example of this particular test exercise.

Region (CUBE)

Region (phones)

Region (Default)

Device Pool main page

Device Pool (CUBE)

Device Pool (phones)

Device Pool (default)

Cisco Unified IP Phone configuration

Note: Notice the "Device Pool" setting is set to "phones". This configuration allows the IP phone calling other local IP phones to use G711 codec, and use G729 codec when the call is made or received out/from the AT&T SIP network (PSTN) as per the region relationship settings.

Cisco Unified IP phone DN configuration

Note: Notice the "External Phone Number Mask", this field is required when configuring end user for use with Meetingplace..

Configuring Application User for Meetingplace application (Optional configuration used for LDAP sync)

Configuring User Group for Meetingplace application (Optional configuration used for LDAP sync)

Configuring End User for Meetingplace application (Optional configuration used for LDAP sync)

IOS conference bridge for G729 conferencing

Note: CUCM requires a conference bridge resource for three-way conferencing that include g729 rtp streams. This CUBE CFB resource is placed in the "CUBE" device pool.

Sample IOS configuration for conference bridge registration to CUCM

voice-card 0

dspfarm

dsp services dspfarm

sccp local GigabitEthernet0/1

sccp ccm 172.X.X.X identifier 1 version 6.0

sccp

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register cfb0018185bb7a1

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 6

associate application SCCP

Transcoder configuration

Note: If your network will support more than one codec flavor, it is recommended to have a transcoder resource on Cisco Unified CM.

Sample IOS gateway configuration for transcoder registration to Cisco Unified CM

voice-card 0

dspfarm

dsp services dspfarm

sccp local GigabitEthernet0/1

sccp ccm 10..X.X;X identifier 1 version 6.0

sccp

sccp ccm group 1

associate ccm 1 priority 1

associate profile 2 register mtp0123456789ab

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

Route Pattern main page

Route Pattern (outgoing call to AT&T)

Route Pattern (outgoing call to AT&T international)

Note: There is no difference between the local outgoing call to AT&T route pattern and the international outgoing call to AT&T route pattern, except for the matching numbering string sequence. This route pattern was included to show that there is no difference device wise or settings wise between local and international route patterns.

Route Pattern (incoming call to Meetingplace)

Multiple Route Pattern are typically deployed (up to 4 can be published) to offer Toll Free, Local CO number and internal dial plan numbers for users to dial into MeetingPlace conferences from any device.

Route Pattern (incoming FAX call)

Note: In the route pattern example above the SIP gateway was set to use G711 as the FAX codec, but both G.711 as well as T.38 codecs were tested successfully. Further in the document you will find how to configure both a G711 SIP trunk and a G729 SIP trunk (upspeed to T.38 during fax call). You would assign the FAX SIP trunk of choice to this route pattern.

DNIS Translation-Pattern

Note: In this translation pattern example the 10-digit DID number incoming from AT&T/CUBE is translated to a 4-digit local directory number.

SIP Gateway (for fax)

T.38 configuration

Note: The device pool is set to "CUBE"

G.711

Note: The device pool is set to "Default".

Enabling PRACK for early-media negotiation

Note: Some PSTN network call prompters utilize early-media cut-through to offer menu options to the caller (DTMF select menu) before the call is connected. In order for CUCM/CUBE solution to achieve successful early-media cut-through the CUCM to CUBE call leg must be enabled with SIP PRACK. To enable SIP PRACK on Cisco Unified CM you must set the parameter "SIP Rel1XXX Enabled" to "True". The parameter is found under SystemèService Parametersè<Server Name or IP address>èCisco CallManager (Active)èClusterwide Parameters (Device-SIP), in the Cisco Unified CM.

Configuring Cisco Unity integration

Voice mail Pilot

Voice mail port

Voice mail profile

Unity Line Group

Unity Hunt Pilot

Unity Hunt List

Configuring MWI

Note: In this configuration DN 1000 was assigned to turn on (light on) MWI indicators on locally registered Cisco IP phones and DN 1001 to turn off (light off). You must ensure that these DN's are correctly matched and assigned to the Cisco Unity system for correct MWI performance

Note: For a complete guide on how to administer Cisco Unified Communications Manager 7.1 go to: Cisco Unified Communications Manager Administration Guide, Release 7.1(2) [Cisco Unified Communications Manager (CallManager)] (http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/bccm-712-cm.html). You can also obtain the CUCM SRND at: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/docguide/7_1_2/dg712.html

Configuring a Cisco IOS SIP Gateway for FAX calls using T.38 or G711

c2801#sh run

Building configuration...

Current configuration : 2765 bytes

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

hostname c2801

boot-start-marker

boot-end-marker

logging buffered 100000000

no logging console

enable password cisco

no aaa new-model

ip cef

multilink bundle-name authenticated

voice-card 0

dsp services dspfarm

voice service voip

h323

sip

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

Configuring Meetingplace

Meetingplace version

Directory service admin

Note: This is the application user created in CUCM

User group

User Profile

SIP configuration

This configuration is used for MeetingPlace "Outdial" calls and directs outbound calls to the UC Manager via SIP trunk. All UC Manager dial rules and CSS should be configured to provide toll fraud restrictions.

Note: IP address of SIP proxy server 1 is the IP address of CUCM server (SIP trunk).

SNMP (public)

Note: You must have correctly matching SNMP community strings between the MeetingPlace Application server and the Cisco Unified MeetingPlace 7.0 Media Server for correct performance.

Media Parameters

Media server viewed from the application server

MeetingPlace Media server (Audio Blade)

Note: For a complete guide on how to administer Cisco Unified MeetingPlace 7.0, including integration procedure to Cisco Unified Communications Manager go to: Configuration Guide for Cisco Unified MeetingPlace Release 7.0 [Cisco Unified MeetingPlace] (http://www.cisco.com/en/US/docs/voice_ip_comm/meetingplace/7x/english/books/admin_guides/configuration_guide_7_0.html)

Configuring Unity to CUCM (SCCP integration)

UTIM configuration

Subscriber configuration

Note: To find detail configuration steps on how to integrate to CUCM using either SCCP or SIP go to: http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_installation_and_configuration_guides_list.html

Acronyms

Acronym	Definitions
SIP	Session Initiation Protocol
MGCP	Media Gateway Control Protocol
SCCP	Skinny Client Control Protocol
CUCM	Cisco Unified Communications Manager
CUBE	Cisco Unified Border Element
SP	Service Provider
PSTN	Public switched telephone network
IP	Internet Protocol
TDM	Time-division multiplexing
CODEC	Coder-Decoder (in this document a device used to digitize and undigitize voice signals)

Appendix A

Configuring redirect number expansion feature using voice translation rules in Cisco IOS

The following IOS command takes a 4-digit extension number and expands it to a 10-digit DID number, but this command is broken in the tested CUBE IOS version 12.4(22)T, CSCsx62600.

voice translation-rule 1

rule 1 // /510555\1/

voice translation-profile expand-redirect

translate redirect-called 1

To test your translation rule use the following command example.

c3825_CUBE# test voice translation-rule 1 1000

Matched with rule 1

Original number: 1000 Translated number: 5105551000

Original number type: none Translated number type: none

Original number plan: none Translated number plan: none

Once you have created your translation rule and placed it into a translation profile where it is used to expand the "redirect-called" number (Diversion header number) you must apply the translation profile to the appropriate dial-peer

Example:

dial-peer voice 1999 voip

translation-profile incoming expand-redirect

destination-pattern 1[1-9,1-9,1-9].......

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 1

voice-class sip early-offer forced

voice-class sip profiles 1

session protocol sipv2

session target ipv4:207.242.225.200

incoming called-number 1..........

dtmf-relay rtp-nte

fax-relay sg3-to-g3

fax rate 14400

fax protocol pass-through g711ulaw

Fir more information on how to configure and apply translation rules go to: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

Troubleshooting DTMF interoperability issues (RFC2833 payload-type value mismatch)

For proper end-to-end transmission of DTMF tones, necessary when accessing Auto Attendant (IVR) or Voice mail applications, the IETF has defined RFC2833 to ensure DTMF interoperability between systems. Both AT&T IP FlexReach services and Cisco Unified Border Element support RFC2833, but there is a caveat as to how Cisco Unified Border Element has implemented RFC2833. This RFC2833 implementation caveat may cause DTMF issues during calls originating from AT&T network towards Cisco Unified CM (PSTNèAT&T IP FlexReachèCisco UC user/application), this is a non-issue for calls originating on the Cisco UC side out towards AT&T IP FlexReach network. This appendix provides a step-by-step procedure of how to troubleshoot and repair any DTMF interoperability issue that may arise between your Cisco Unified Communications Manager/Cisco Unified Border Element solution and AT&T IP FlexReach services. AT&T's IP FlexReach services has the ability to assign RTP payload-type value dynamically in order to transport DTMF payloads over RTP (RFC2833), although Cisco Unified Border Element is capable of supporting any value within the dynamically assignable range (96-127) Cisco Unified Border Element can not dynamically assign the payload-type value. The Cisco Unified Border Element dial-peer configuration must be configured (hardset) to the payload-type value expected on the incoming dial-peer.

Example:

dial-peer voice 732320 voip

description dial-peer to Cisco Unified CM

destination-pattern 732320....

signaling forward unconditional

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 1

voice-class sip early-offer forced

session protocol sipv2

session target ipv4:yy.yy.yy.yy (Cisco Unified CM IP address)

dtmf-relay rtp-nte

fax rate 14400

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

fax-relay sg3-to-g3

For this reason AT&T has taken a best effort approach to harmonize all SIP systems within their IP FlexReach service to utilize RTP payload-type value "100" and that is why this application note utilizes value 100 in its dial-peer configuration examples. But, because AT&T cannot guarantee a RTP payload-type value of "100" will be offered in all provisioning instances you can enable SIP debug commands on the Cisco Unified Border Element to obtain the RTP payload-type value being offered within the initial SIP INVITE message sent by AT&T on incoming calls. Please follow instructions below.

Step 1:

telnet or connect to the CUBE console and configure SIP messages trace capture

Router>

Enter privileged EXEC mode

Router>enable

Router#

Enter global config mode

Router# config t

Router(config)#

If telnetted, set logging buffer for debug capture and allocate sufficient buffer size to capture SIP messages

Router(config)# logging buffered debug <enter>

Router(config)# logging buffered 10000000 <enter>

If consoled, set logging console and buffer size

Router(config)# logging console<enter>

Router(config)# logging buffered 10000000 <enter>

Enable SIP messages debug trace capture

Router# debug ccsip messages<enter>

SIP Call messages tracing is enabled

Step 2:

Place a call from the PSTN dialing your AT&T IP FlexReach provisioned DID number in order for the call to be directed to your Cisco Unified Border Element (CUBE). At this point your CUBE would have received the call and captured the related SIP messages.

Step 3:

Perform a "show logging" command to print out the SIP message trace on the CUBE CLI prompt.

Example:

Router# show logging

Syslog logging: enabled (0 messages dropped, 2 messages rate-limited,

0 flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.

No Inactive Message Discriminator.

Console logging: disabled

Monitor logging: disabled

Buffer logging: level debugging, 1816 messages logged, xml disabled,

filtering disabled

Logging Exception size (4096 bytes)

Count and timestamp logging messages: disabled

Persistent logging: disabled

No active filter modules.

ESM: 0 messages dropped

Trap logging: level informational, 56 message lines logged

Log Buffer (100000000 bytes):

*Jul 1 23:23:47.387: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:7323204065@x.x.x.x:5060 SIP/2.0

Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK2rnefs20d80h0dc95681.1

From: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

To: <sip:7323204065@x.x.x.x;user=phone>

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

CSeq: 1 INVITE

Max-Forwards: 68

Contact: <sip:+15108280583@y.y.y.y:5060;transport=udp>

P-Asserted-Identity: "OUT_OF_AREA" <sip:5108280583@y.y.y.y:5060>

P-DCS-Billing-Info: CC152DB5003D6BD400000000312D303530303030000058A7/0@z.z.z.z

Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed

Accept-Language: en; q=0.0

Content-Length: 268

Content-Disposition: session; handling=required

Content-Type: application/sdp

v=0

o=Sonus_UAC 26532 12960 IN IP4 y.y.y.y

s=SIP Media Capabilities

c=IN IP4 y.y.y.y

t=0 0

m=audio 26798 RTP/AVP 18 0 100

a=rtpmap:18 G729/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=sendrecv

a=maxptime:20

*Jul 1 23:23:47.395: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK2rnefs20d80h0dc95681.1

From: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

To: <sip:7323204065@x.x.x.x;user=phone>

Date: Tue, 01 Jul 2008 23:23:47 GMT

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow-Events: telephone-event

Content-Length: 0

*Jul 1 23:23:47.399: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:7323204065@172.20.110.254:5060 SIP/2.0

Via: SIP/2.0/UDP 172.20.110.1:5060;branch=z9hG4bK7F1734

From: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

To: <sip:7323204065@172.20.110.254>

Date: Tue, 01 Jul 2008 23:23:47 GMT

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Cisco-Guid: 2532160513-1190859229-2166670528-1526950391

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1214954627

Contact: <sip:+15108280583@172.20.110.1:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 67

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 307

v=0

o=CiscoSystemsSIP-GW-UserAgent 1097 6859 IN IP4 172.20.110.1

s=SIP Call

c=IN IP4 172.20.110.1

t=0 0

m=audio 16406 RTP/AVP 18 0 100 19

c=IN IP4 172.20.110.1

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:0 PCMU/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=rtpmap:19 CN/8000

*Jul 1 23:23:47.611: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Date: Tue, 01 Jul 2008 22:19:22 GMT

From: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

Allow-Events: presence

Content-Length: 0

To: <sip:7323204065@172.20.110.254>

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Via: SIP/2.0/UDP 172.20.110.1:5060;branch=z9hG4bK7F1734

CSeq: 101 INVITE

*Jul 1 23:23:47.911: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Date: Tue, 01 Jul 2008 22:19:22 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH

From: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

Allow-Events: presence

Remote-Party-ID: "UCM61-4065-" <sip:4065@172.20.110.254>;party=called;screen=yes;privacy=off

Content-Length: 0

To: <sip:7323204065@172.20.110.254>;tag=4ba3f703-6e57-4aa0-a9a5-4fbc7983fad3-20304512

Contact: <sip:7323204065@172.20.110.254:5060>

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Via: SIP/2.0/UDP 172.20.110.1:5060;branch=z9hG4bK7F1734

CSeq: 101 INVITE

*Jul 1 23:23:47.911: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK2rnefs20d80h0dc95681.1

From: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

To: <sip:7323204065@x.x.x.x;user=phone>;tag=7239B138-19B0

Date: Tue, 01 Jul 2008 23:23:47 GMT

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <sip:7323204065@x.x.x.x:5060>

Content-Length: 0

*Jul 1 23:23:53.387: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Date: Tue, 01 Jul 2008 22:19:22 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH

From: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

Allow-Events: presence

Supported: replaces

Remote-Party-ID: "UCM61-4065-" <sip:4065@172.20.110.254>;party=called;screen=yes;privacy=off

Content-Length: 218

To: <sip:7323204065@172.20.110.254>;tag=4ba3f703-6e57-4aa0-a9a5-4fbc7983fad3-20304512

Contact: <sip:7323204065@172.20.110.254:5060>

Content-Type: application/sdp

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Via: SIP/2.0/UDP 172.20.110.1:5060;branch=z9hG4bK7F1734

CSeq: 101 INVITE

v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.20.110.254

s=SIP Call

c=IN IP4 172.20.110.196

t=0 0

m=audio 29400 RTP/AVP 18 100

a=rtpmap:18 G729/8000

a=ptime:20

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

*Jul 1 23:23:53.387: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:7323204065@172.20.110.254:5060 SIP/2.0

Via: SIP/2.0/UDP 172.20.110.1:5060;branch=z9hG4bK80246F

From: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

To: <sip:7323204065@172.20.110.254>;tag=4ba3f703-6e57-4aa0-a9a5-4fbc7983fad3-20304512

Date: Tue, 01 Jul 2008 23:23:47 GMT

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jul 1 23:23:53.391: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK2rnefs20d80h0dc95681.1

From: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

To: <sip:7323204065@x.x.x.x;user=phone>;tag=7239B138-19B0

Date: Tue, 01 Jul 2008 23:23:47 GMT

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <sip:7323204065@x.x.x.x:5060>

Supported: replaces

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 274

v=0

o=CiscoSystemsSIP-GW-UserAgent 9758 2556 IN IP4 x.x.x.x

s=SIP Call

c=IN IP4 x.x.x.x

t=0 0

m=audio 19416 RTP/AVP 18 100

c=IN IP4 x.x.x.x

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-16

a=ptime:20

*Jul 1 23:23:53.555: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:7323204065@x.x.x.x:5060 SIP/2.0

Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK2juhh830187g2dc9u641.1

Max-Forwards: 69

To: <sip:7323204065@x.x.x.x;user=phone>;tag=7239B138-19B0

From: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

CSeq: 1 ACK

Content-Length: 0

Contact: <sip:+15108280583@y.y.y.y:5060;transport=udp>

*Jul 1 23:23:56.263: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:+15108280583@172.20.110.1:5060 SIP/2.0

Date: Tue, 01 Jul 2008 22:19:28 GMT

From: <sip:7323204065@172.20.110.254>;tag=4ba3f703-6e57-4aa0-a9a5-4fbc7983fad3-20304512

Content-Length: 0

User-Agent: Cisco-CUCM6.1

To: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Via: SIP/2.0/UDP 172.20.110.254:5060;branch=z9hG4bK5e563b6bd

CSeq: 101 BYE

Max-Forwards: 70

*Jul 1 23:23:56.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.20.110.254:5060;branch=z9hG4bK5e563b6bd

From: <sip:7323204065@172.20.110.254>;tag=4ba3f703-6e57-4aa0-a9a5-4fbc7983fad3-20304512

To: "OUT_OF_AREA" <sip:+15108280583@172.20.110.1>;tag=7239AF38-1ED

Date: Tue, 01 Jul 2008 23:23:56 GMT

Call-ID: 96EF8879-46FB11DD-812AC4C0-5B0369F7@172.20.110.1

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 BYE

Reason: Q.850;cause=16

Content-Length: 0

*Jul 1 23:23:56.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:+15108280583@y.y.y.y:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK813F2

From: <sip:7323204065@x.x.x.x;user=phone>;tag=7239B138-19B0

To: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

Date: Tue, 01 Jul 2008 23:23:53 GMT

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1214954636

CSeq: 101 BYE

Reason: Q.850;cause=16

Content-Length: 0

*Jul 1 23:23:56.395: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK813F2

From: <sip:7323204065@x.x.x.x;user=phone>;tag=7239B138-19B0

To: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

Timestamp: 1214954636

CSeq: 101 BYE

Content-Length: 0

Contact: <sip:+15108280583@y.y.y.y:5060;transport=udp>

Step 4:

Obtaining the RTP payload-value for DTMF transmission and configuring the payload-type value in the appropriate dial-peer.

The rtp payload-value will be obtained by reading the initial incoming SIP INVITE and will be located within the SDP offer.

Example:

Received:

INVITE sip:7323204065@x.x.x.x:5060 SIP/2.0

Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK2rnefs20d80h0dc95681.1

From: "OUT_OF_AREA" <sip:+15108280583@y.y.y.y:5060;user=phone>;tag=dsa78ed2f0

To: <sip:7323204065@x.x.x.x;user=phone>

Call-ID: ASE_1214951256345_5608_null_z.z.z.z

CSeq: 1 INVITE

Max-Forwards: 68

Contact: <sip:+15108280583@y.y.y.y:5060;transport=udp>

P-Asserted-Identity: "OUT_OF_AREA" <sip:5108280583@y.y.y.y:5060>

P-DCS-Billing-Info: CC152DB5003D6BD400000000312D303530303030000058A7/0@a.a.a.a

Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed

Accept-Language: en; q=0.0

Content-Length: 268

Content-Disposition: session; handling=required

Content-Type: application/sdp

v=0

o=Sonus_UAC 26532 12960 IN IP4 y.y.y.y

s=SIP Media Capabilities

c=IN IP4 y.y.y.y

t=0 0

m=audio 26798 RTP/AVP 18 0 96

a=rtpmap:18 G729/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:96 telephone-event/8000 è In this example rtp payload-type for telephone-event (RFC2833) is "96"

a=fmtp:96 0-15 è 0-15 is related to DTMF characters 0-9, *, #, A-D

a=sendrecv

a=maxptime:20

Once you have obtained the RTP payload-value you will need to configure this newly acquired value into all appropriate dial-peers

Example:

Originally configured dial-peer values, as per application note CUBE configuration section

Router# show running-config | begin dial-peer voice 4060

dial-peer voice 732320 voip

description Outgoing dial-peer to Cisco Unified CM

destination-pattern 732320....

signaling forward unconditional

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 1

voice-class sip early-offer forced

session protocol sipv2

session target ipv4:yy.yy.yy.yy

incoming called-number 732320....

dtmf-relay rtp-nte

fax rate 14400

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

fax-relay sg3-to-g3

Router# config t

Router(config)#dial-peer voice 732320

Note: Cisco IOS assigns hardcoded RTP payload-type values to all IOS supported codecs. In the example above RFC2833 is being offered on rtp payload-type value 96 but IOS, by default, assigns value 96 to Cisco codec "cisco-codec-fax-ind". You must change the RTP payload-value assignment of "cisco-codec-fax-ind" codec to a different value in order to assign value 96 to payload-type "nte" (RFC2833). You may face this situation with other rtp payload-type values. Refer to table below and link for all IOS default payload-type values.

nse number	A named signaling event (NSE). Range: 96 to 117. Default: 100.
nte number	A named telephone event (NTE). Range: 96 to 127. Default: 101.
cisco-codec-fax-ind number	Cisco codec fax indication. Range: 96 to 127. Default: 96.

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_r1.html#wp1571622

Router(config-dial-peer)#rtp payload-type cisco-codec-fax-ind 98 èThis frees up value 96

Router(config-dial-peer)# rtp payload-type nte 96 è This assigns value 96 to RFC2833

Router(config-dial-peer)# default pyaload-type nse è This sets nse value back to default

New dial-peer:

Router# show running-config | begin dial-peer voice 4060

dial-peer voice 732320 voip

description Outgoing dial-peer to Cisco Unified CM

destination-pattern 732320....

signaling forward unconditional

rtp payload-type cisco-codec-fax-ind 98

rtp payload-type nte 96

voice-class codec 1

voice-class sip early-offer forced

session protocol sipv2

session target ipv4:yy.yy.yy.yy

incoming called-number 732320....

dtmf-relay rtp-nte

fax rate 14400

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

fax-relay sg3-to-g3

CUCM Cluster configuration with a single Cisco Unified Border Element

To support signaling from multiple UCM Servers, within a UCM Cluster, additional dial-peers can be added to the CUBE dial plan. Each additional dial peer supporting signaling connectivity to the additional server(s).

Two dial peers are utilized in this configuration guide. Dial-peer 1999 faces the AT&T Flexible Reach service and dial-peer 732320 faces the UCM server.

Using the CUBE configuration referenced on page 9 of this guide as an example, create new CUBE dial peers for each server/trunk within the UCM Cluster.

Publisher

dial-peer voice 732320 voip

description Outgoing dial-peer to Cisco Unified CM pri

preference 0

destination-pattern 732320....

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 1

voice-class sip profiles 1

session protocol sipv2

session target ipv4:1.1.1.1

dtmf-relay rtp-nte

fax rate 14400

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

fax-relay sg3-to-g3

Subscriber

dial-peer voice 732321 voip

description Outgoing dial-peer to Cisco Unified CM sub

preference 1²⁹

destination-pattern 732320....

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 1

voice-class sip profiles 1

session protocol sipv2

session target ipv4:1.1.1.2

dtmf-relay rtp-nte

fax rate 14400

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

fax-relay sg3-to-g3

Incoming call to CUCM dial-peer

dial-peer voice 732322 voip

description incoming call to CUCM dial-peer

rtp payload-type nse 99

rtp payload-type nte 100

voice-class codec 1

voice-class sip profiles 1

session protocol sipv2

incoming called-number 732320....

dtmf-relay rtp-nte

fax rate 14400

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

fax-relay sg3-to-g3

A correct dial plan configuration is required for load balancing across the Cisco UCM servers, within the cluster and in server failure scenarios.

Information on configuring a more enhanced dial plan can be found on cisco.com

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.
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Printed in the USA

¹This command enables DSP farming, allowing DSP resources to register to Cisco Unified CM as MTP, CFB or Transcoder devices

²Enables IP addressing hiding between the private network (CUCM side) and the public network (AT&T IP Flex-reach side)

³This command enables CUBEs basic IP-to-IP voice communication feature.

⁴This command enables T.38 fax at a global level, meaning all VoIP dial-peers not configured for a specific fax protocol will utilize this setting. If T.38 protocol should be applied to indvidual dial-peers only this command must be disabled using the "no" form of the command and configure the command under the appropriate dial-peers

⁵This command allows for SIP error messages to pass-through end-to-end without modification through CUBE.

⁶The asserted-id pai command enables the delivery of caller id information using P-asserted-ID method, across the SIP trunk, on CUBE. See caveat section for configuration requirements information of this command.

⁷This command must be enabled at a global level to maintain integrity of SIP signaling between AT&T network and Cisco Unified CM across CUBE.

⁸This command allows for privacy settings to be transparently passed across between AT&T network and Cisco Unified CM. The command can be set either at a global level, as it is in this example, or it can be set at the dial-peer level.

⁹This command allows CUBE to negotiate all flavors of G729 codec and must be configured in order to interoperate seamlessly across AT&T's BVOIP services. The command can be enabled either globally, such as in this example, or per dial-peer basis using the "voice-class sip g729 annexb-all" command.

¹⁰This command enables multiple codec support and performs codec filtering required for correct interoperability between AT&T SIP network and Cisco Unified CM.

¹¹This SIP profile expands the Diversion header number from a 4-digit extension to a full 10-digit DID number in order to obtain interoperability with AT&T's HIPCS (NSN) served users during call-forward calls.

¹²These commands configure the shared DSP resources as conference bridge (CFB) device for Cisco Unified CM

¹³To configure a redundant outgoing dial-peer towards a second (backup) AT&T border element you will create a second dial-peer with the same config values except you will set the :session target IP address for the AT&T back border element and you will need to set the "preference" command under both dial-peers as appropriate.

¹⁴This command frees payload type value 100, in order to assign it to RFC2833 application (nte)

¹⁵This command assigns payload type value 100 to nte (RFC2833) application

¹⁶Assigns voice class codec 1 settings to dial-peer (codec support and filtering)

¹⁷ This command allows for privacy settings to be transparently passed across between AT&T network and Cisco Unified CM. In this example the command is set at the dial-peer level, you can also set the command at a global level to effect all dial-peers without needing to set the command on each dial-peer.

¹⁸This command enables delay offer-to-early offer conversion of initial SIP INVITE message to calls matched to this dial-peer level

¹⁹This command enables the SIP profiles feature for calls matching this dial-peer

²⁰This command sets the SIP server target for outgoing SIP calls

²¹This command assigns configured dial-peer properties to incoming calls matching called number string

²²This command enables DTMF digit passing using RTP NTE (RFC2833) to calls matching this dial-peer.

²³Example of how to configure T.38 fax on a per dial-peer basis

²⁴This command forces the configured nte payload type value ( in this case 100 value) for incoming calls to CUCM

²⁵Full DID number to reach Meetingplace from PSTN

²⁶This command forces the configured nte payload type value ( in this case 101 default value) for incoming calls to Meetingplace

²⁷This command enables NTE RTP for DTMF digit passing, and the statement :digit-drop" ensures in-band DTMF tones are not carried toward Meetingplace server.

²⁸This command should be enabled when failover dial-peers are configured (for example: when using a CUCM cluster). See appendix A for details on dial-peer configuration when using a CUCM cluster.

²⁹The "preference" command is used to prioritize dial-peers with matching destination-patterns. You can also prioritize dial-peers with matching destination patterns by order of configuration entry to the CLI, be aware that if you need to set high priority to a newly added dial-peer you will need to use the "preference" command.

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