application note at&t ip flexreach: connecting cisco ... 7.1 via the cisco unified ... 7.1 with...

© 2009 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

of 125 EDCS# 817180 Rev # 5

Application Note

AT&T IP FlexReach: Connecting Cisco Unified Communications Manager 7.1 via the Cisco Unified Border Element 1.4 using SIP

Oct 2, 2009

Table of Table of Table of Table of ContentsContentsContentsContents

Introduction..............................................................................................................................................................................................................2 Network Topology....................................................................................................................................................................................................3 System Components.................................................................................................................................................................................................4

Hardware Components........................................................................................................................................................................................4 Software Requirements........................................................................................................................................................................................4

Features....................................................................................................................................................................................................................5 Features Newly Supported in CISCO UBE 1.4...................................................................................................................................................5 Features Supported..............................................................................................................................................................................................5 Features Not Supported.......................................................................................................................................................................................5

AT&T Flexreach E911 Service Notation:.................................................................................................................................................................5 Caveats.....................................................................................................................................................................................................................6 Configuration............................................................................................................................................................................................................7

Cisco IOS version................................................................................................................................................................................................7 Configuring Cisco Unified Border Element (Cisco UBE)...................................................................................................................................8 Configuring the Cisco Unified Communications Manager ...............................................................................................................................14 Configuring a Cisco IOS SIP Gateway for FAX calls using T.38.....................................................................................................................66 How to configure MGCP or H323 IOS gateway for fax T.38...........................................................................................................................68 Configuring Meetingplace.................................................................................................................................................................................85 Configuring Cisco Unity Connection using SIP..............................................................................................................................................105

Acronyms..............................................................................................................................................................................................................120 Appendix A..........................................................................................................................................................................................................121

Configuring redirect number expansion feature using voice translation rules in Cisco IOS...........................................................................121 Cisco Unified Border Element web link references.........................................................................................................................................122 CISCO UCM Cluster configuration with a single Cisco Unified Border Element..........................................................................................122

© 2009 Cisco Systems, Inc. All rights reserved.

Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com of 125

IntroductionIntroductionIntroductionIntroduction

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 (CISCO UCM) 7.1 with a Cisco Unified Border Element (CISCO UBE) 1.4 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 Connection 7.1 messaging integrated to the Cisco Unified Communications Manager. The deployment model covered in this application note is Customer Premises Equipment (Unity/Meetingplace/CISCO UCM/CISCO UBE) 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.

• 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 Connection), auto-attendant (Cisco Unity), fax using T.38 and G.711 (G3 and SG3 speeds), teleconferencing, failover of unresponsive SIP network to PSTN 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. Care must be taken by the network administrator deploying CISCO UBE to ensure these commands are set per each dial-peer required, 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/Cisco UCM/admin/7_1_2/ccmcfg/bccm-712-cm.html)

http://tools.cisco.com/search/display?url=http%3A%2F%2Fwww.cisco.com%2Fen%2FUS%2Fdocs%2Fvoice_ip_comm%2Fcucm%2Fadmin%2F7_1_2%2Fccmcfg%2Fbccm-712-cm.html&pos=3&strqueryid=5&websessionid=CjZKlk075XEQG3Ybhrpw8Aa





Network TopologyNetwork TopologyNetwork TopologyNetwork Topology

Figure 1. Basic Call Setup

• The CISCO UBE 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.



System ComponentsSystem ComponentsSystem ComponentsSystem Components

Hardware Components

• Cisco IOS gateway running CISCO UBE 1.4 (IOS version 15.0.1XA and future 15.T releases)

• 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 CISCO UBE 1.4 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/cgi-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 Connection)

• 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. Please refer to the IOS fax gateway configuration section for details.

Software Requirements

• Cisco Unified CM 7.1 and later 7.x releases. This solution was tested with 7.1.2.21900-5

• CISCO UBE version 1.4 IOS version 15.0.1XA and later 15.T releases. This configuration was tested with IP VOICE (ipvoicek9-mz.)

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

• The documented CISCO UBE 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 Connection 7.1.2.39000-61 software

• Consult your Cisco representative for the correct IOS image and for the specific application and Device Unit License and Feature License requirements for all your Cisco Unified Communications components.

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

http://www.cisco.com/cgi-bin/Support/DSP/cisco_dsp_calc.pl

http://www.cisco.com/cgi-bin/Software/Iosplanner/Planner-tool/iosresult.cgi?get_crypto=&data_from=&hardware_name=2821&software_name=INT%20VOICE%2FVIDEO%2C%20IPIP%20GW%2C%20TDMIP%20GW&release_name=12.4.20T&majorRel=12.4&state=:RL:HW:SW&type=Early%20Deployment

http://www.cisco.com/cgi-bin/Software/Iosplanner/Planner-tool/iosplanner.cgi?get_crypto=&data_from=&hardware_name=&software_name=INT%20VOICE%2FVIDEO%2C%20IPIPGW%2C%20TDMIP%20GW%20AES&release_name=12.4.20T&majorRel=12.4&state=:RL:SW&type=Early%20Deployment

http://www.cisco.com/cgi-bin/Software/Iosplanner/Planner-tool/iosplanner.cgi?get_crypto=&data_from=&hardware_name=&software_name=INT%20VOICE%2FVIDEO%2C%20IPIPGW%2C%20TDMIP%20GW%20AES&release_name=12.4.20T&majorRel=12.4&state=:RL:SW&type=Early%20Deployment



FeaturesFeaturesFeaturesFeatures

Features Newly Supported in CISCO UBE 1.4

• Failover to PSTN. The Cisco Unified Border Element can be configured to fallback to a legacy PSTN connection (e.g. T1-PRI) when all attempts to route a call using SIP trunk fail. See configuration section for details.

• Cisco Unified Border Element version 1.4 can perform payload-type value interworking for RFC2833 DTMF packets when the RTP payload-type values are mismatched between the inbound and outbound SIP call legs.

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

• Fax over G.711 (See Caveat section for details)

• Incoming DNIS Translation and Routing

• CISCO UBE: performs Delayed-Offer-to-Early-Offer conversion of an initial SIP INVITE 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)

• Failover (From non-responsive SIP network to legacy PSTN circuit)

Features Not Supported

• CISCO UCM/CISCO UBE Codec negotiation of G.726

AT&T Flexreach AT&T Flexreach AT&T Flexreach AT&T Flexreach E911 E911 E911 E911 Service Service Service Service NotationNotationNotationNotation::::

• While AT&T IP Flexible Reach services supports 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.



CaveatsCaveatsCaveatsCaveats

Resolved CaveatsResolved CaveatsResolved CaveatsResolved Caveats

• Payload Type - prior versions of CISCO UBE required static dial peer configuration to set a payload-type value used for digit passing

with RFC2833. This has been resolved in CISCO UBE 1.4. See “Features Newly Supported” and configuration section for details.

New Caveats

• Fax Pass-through: Upspeed to G.711 is not supported with Cisco Unified Communications Manager. If T.38 is not an option Fax over G.711 will be required. Deploying Fax over G.711 will require specific dial peer configurations that apply to the fax GW and Cisco Unified Border Element. See the Cisco Fax Configuration section for details.

Configuration considerations

• 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 CISCO UBE 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 CISCO UBE 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 forwarded calls from CISCO UCM 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 CISCO UCM translation pattern. Because we use 4-digit extensions on our CISCO UCM 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 CISCO UBE (See configuration 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.

• Upon receiving inbound calls, AT&T SIP network will always select the first choice codec presented in the initial SIP INVITE (unless the end-device does not support the listed preferred codec), and processes calls accordingly. Customers wishing to place G.711-only calls must configure separate dial-peer(s) on CISCO UBE with G.711 codec assigned. This dial-peer is matched on Cisco Unified CM by a Route Pattern directing calls to it. Typically, this solution is used for fax/modem transmissions using G711. To ensure that inbound fax/modem calls are established using G.711 codec, assign SIP trunks and/or MGCP/H.323 gateways supporting fax/modem into Cisco Unified CM Region(s) configured to use G.711 codec.



CCCConfigurationonfigurationonfigurationonfiguration

Cisco IOS version

c3825_CUBE#sh ver Cisco IOS Software, 3800 Software (C3825-IPVOICEK9-M), Version 15.0(1)XA, RELEAS E SOFTWARE (fc2) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2009 by Cisco Systems, Inc. Compiled Thu 22-Oct-09 01:41 by prod_rel_team ROM: System Bootstrap, Version 12.3(11r)T2, RELEASE SOFTWARE (fc1) c3825_CUBE uptime is 24 minutes System returned to ROM by reload at 01:46:30 UTC Fri Nov 6 2009 System restarted at 01:49:07 UTC Fri Nov 6 2009 System image file is "flash:c3825-ipvoicek9-mz.150-1.XA.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 [email protected]. Cisco 3825 (revision 1.1) with 489471K/34816K bytes of memory. Processor board ID FTX1025A25D 2 Gigabit Ethernet interfaces 5 Serial interfaces 2 Channelized T1/PRI ports DRAM configuration is 64 bits wide with parity enabled. 479K bytes of NVRAM. 62720K bytes of ATA System CompactFlash (Read/Write) License Info: License UDI: ------------------------------------------------- Device# PID SN ------------------------------------------------- *0 CISCO3825 FTX1025A25D Configuration register is 0x2102

http://www.cisco.com/techsupport

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

mailto:[email protected]



Configuring Cisco Unified Border Element (Cisco UBE)

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 ! card type t1 0 2 logging message-counter syslog logging buffered 100000000 ! no aaa new-model network-clock-participate wic 2 network-clock-select 1 T1 0/2/0 ! dot11 syslog ip source-route ip cef ! ! ! ! no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm1 ! ! ! voice service voip address-hiding2 allow-connections sip to sip3 redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none4

1 This command enables DSP farming, allowing DSP resources to register to Cisco Unified CM as MTP, CFB or Transcoder devices 2 Enables IP addressing hiding between the private network (CISCO UCM side) and the public network (AT&T IP Flex-reach side) 3 This command enables CUBEs basic IP-to-IP voice communication feature.



h323 sip header-passing error-passthru5 midcall-signaling passthru6 privacy-policy passthru7

no update-callerid g729 annexb-all8 ! ! ! ! ! ! voice class codec 19 codec preference 1 g729r8 ====� codec preference 2 g711ulaw ! ! ! ! ! voice class sip-profiles 1 request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:732320\1@\2>"10 request REINVITE sdp-header Attribute modify "a=T38 FaxFillBitRemoval:0" "" 11 ! ! ! ! ! ! ! ! ! ! ! archive log config hidekeys ! !

4 This 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 individual dial-peers only this command must be disabled using the “no” form of the command and configure the command under the appropriate dial-peers 5 This command allows for SIP error messages to pass-through end-to-end without modification through CUBE. 6 This command must be enabled at a global level to maintain integrity of SIP signaling between AT&T network and Cisco Unified CM across CUBE. 7 This 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. 8 This 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. 9 This command enables multiple codec support and performs codec filtering required for correct interoperability between AT&T SIP network and Cisco Unified CM. 10 This 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. 11 This SIP profile removes the SDP attribute “T38FaxFillBitRemoval:0” from a Cisco IOS gateway upspeed Re-INVITE (inbound call to CPE). Some SIP components within AT&T’s SIP core do not support the “:0” as the boolean value, instead some AT&T devices interpret the full attribute as the boolean value (1=attribute present, 0=attribute not present). For this reason we remove the attribute completely to achieve fax t.38 interoperability across AT&T’s entire SIP core.

This command also controls the payload packet size.Use “codec preference 1 g729r8 bytes 30” to set your payload packet size rate to 30 bytes. Make sure you match the packet size rate set on CISCO UBE to the packet size rate set on CISCO UCM, see CISCO UCM config for details, click on the blue arrow c3825_CUBE(config-class)#codec preference 1 g729r8 bytes ? Each codec sample produces 10 bytes of voice payload. Valid sizes are: 10, 20, 30, 40, 50, 60, 70, 80, 90, 100, 110, 120, 130, 140, 150, 160, 170, 180, 190, 200, 210, 220, 230, 240. Any other value within the range will be rounded down to nearest valid size. <10-244> Choose a voice payload size from the list above



controller T1 0/2/0 pri-group timeslots 1-24 ! ! ! ! interface GigabitEthernet0/0 description outside interface ip address 99.xx.xx.xx 255.255.255.0 ip access-group 101 in duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 description inside interface ip address 192.yy.yy.yy 255.255.255.0 duplex auto speed auto media-type rj45 ! interface Serial0/2/0:23 description T1 ISDN port no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! ip forward-protocol nd ! ! ip route 0.0.0.0 0.0.0.0 99.X.X.1 ip route 192.X.X.0 255.255.255.0 GigabitEthernet0/1 ip http server no ip http secure-server ! access-list 101 permit udp host 207.Y.Y.Y any access-list 101 deny udp any any eq 5060 access-list 101 deny udp any any eq tftp access-list 101 deny tcp any any eq www access-list 101 deny tcp any any eq telnet access-list 101 deny tcp any any eq ftp access-list 101 permit ip any any ! ! ! ! control-plane ! ! voice-port 0/2/0:23 ! ! ! ! sccp local GigabitEthernet0/112 sccp ccm 192.X.X.X identifier 1 version 7.0 12 These sccp commands configure the shared DSP resources as conference bridge (CFB) and as transcoder device for Cisco Unified CM



sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 2 register mtp0123456789ab associate profile 1 register cfb0018185bb7a1 ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 6 associate application SCCP ! ! ! ! ! ! dial-peer voice 1999 voip description Outgoing to AT&T destination-pattern 1[1-9,1-9,1-9]....... voice-class codec 113 voice-class sip asserted-id pai14 voice-class sip privacy-policy passthru15 voice-class sip early-offer forced16 voice-class sip profiles 117 session protocol sipv2 session target ipv4:207.Y.Y.Y18 incoming called-number 1..........19 dtmf-relay rtp-nte 20 fax-relay sg3-to-g3 fax rate 14400 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none21 �� 13 Assigns voice class codec 1 settings to dial-peer (codec support and filtering) 14 The asserted-id pai command enables the delivery of caller id information using P-asserted-ID method, across the SIP trunk, on CUBE. This command can be enabled at the global level under “voice service voip” to affect all SIP dial-peers or under a specific dial-peer to only affect the dial-peer with the command configured 15 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 affect all dial-peers without needing to set the command on each dial-peer. 16 This command enables delay offer-to-early offer conversion of initial SIP INVITE message to calls matched to this dial-peer level 17 This command enables the SIP profiles feature for calls matching this dial-peer 18 This command sets the SIP server target for outgoing SIP calls 19 This command assigns configured dial-peer properties to incoming calls matching called number string 20 This command enables DTMF digit passing using RTP NTE (RFC2833) to calls matching this dial-peer.

If your area does not support fax using T.38 refer to the “Fax over G711 configuration section.



! dial-peer voice 19991 voip22 description Outgoing to AT&T backup SIP preference 123 destination-pattern 1[1-9,1-9,1-9]....... voice-class codec 1 voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip profiles 1 session protocol sipv2 session target ipv4:207.Y.Y.Z incoming called-number 1.......... dtmf-relay rtp-nte fax-relay sg3-to-g3 fax rate 14400 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none ! dial-peer voice 19993 pots description Outgoing to AT&T backup PSTN preference 224 destination-pattern 1[1-9,1-9,1-9]....... direct-inward-dial port 0/2/0:23 forward-digits all ! dial-peer voice 11 voip description Outgoing to AT&T international destination-pattern 011T session protocol sipv2 session target ipv4:207.Y.Y.Y incoming called-number 011T voice-class codec 1 voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip profiles 1 dtmf-relay rtp-nte fax-relay sg3-to-g3 fax rate 14400 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none ! dial-peer voice 732111 voip description Incoming to CISCO UCM destination-pattern 732….... voice-class codec 1 voice-class sip asserted-id pai voice-class sip privacy-policy passthru session protocol sipv2 session target ipv4:192.X.X.X incoming called-number 732320.... dtmf-relay rtp-nte

21 Example of how to configure T.38 fax on a per dial-peer basis 22 This is a fallback dial-peer pointing to a second AT&T border element in case the primary AT&T border element becomes unresponsive to CUBE SIP INVITEs 23 This preference command give 2nd level priority to the dial-peer when a 1xxx-xxx-xxxx number is dialed, 1st priority is dial-peer tag 1999. 24 This preference command give 3rd level priority to the dial-peer when a 1xxx-xxx-xxxx number is dialed, 2nd priority is dial-peer tag 19991.



fax-relay sg3-to-g3 fax rate 14400 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none ! ! ! ! ! ! ! ! ! ! ! ! sip-ua retry invite 2 25 no remote-party-id ! ! ! gatekeeper shutdown ! ! line con 0 password cisco login line aux 0 line vty 0 4 exec-timeout 0 0 password cisco login ! exception data-corruption buffer truncate scheduler allocate 20000 1000 end

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



Configuring the Cisco Unified Communications Manager

Cisco Unified Communications Manager version



Single g729 SIP trunk configuration (Title page)



Single g729 SIP Trunk to AT&T (CISCO UBE) configuration



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

Note: For these particular SIP trunk configurations and applications, the CISCO UCM 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 CISCO UCM 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: Ensure destination IP address is CISCO UBE’s private side IP address to where outgoing calls should be routed



G711 SIP trunk



Note: Ensure destination IP address is CISCO UBE’s private side IP address



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. When configuring the dual SIP trunk to support outbound G711 and G729 calls you must provision the G711 trunk as the primary trunk in the route group, as shown in this config. Also, make sure the route group "Distribution Algorithm"is set to "Top Down”. Once you create the Route Group and Route List and you assign the Route List to the appropriate route pattern(s) you will need to reset the G729 SIP trunk "only" under Device==>Trunk in order for incoming calls to land using G729 if the CUCM end-point supports it. Any time the Route List is reset you will need to individually reset the G729 SIP trunk associated to the route list individually for inbound calls to land g729 when supported.



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



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: CISCO UCM requires a conference bridge resource for three-way conferencing that include g729 rtp streams. This CISCO UBE CFB resource is placed in the “CISCO UBE” device pool.

Sample IOS configuration for conference bridge registration to CISCO UCM

voice-card 0 dspfarm dsp services dspfarm ! sccp local GigabitEthernet0/1 sccp ccm 192.X.X.X identifier 1 version 7.0 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/CISCO UBE is translated to a 4-digit local directory number.



SIP Gateway (for fax)

T.38 configuration

Note: The device pool is set to “CUBE”



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 CISCO UCM/CISCO UBE solution to achieve successful early-media cut-through the CISCO UCM to CISCO UBE 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 System�Service Parameters�<Server Name or IP address>�Cisco CallManager (Active)�Clusterwide Parameters (Device-SIP), in the Cisco Unified CM.



Configuring codec payload (packet) size

Configuring Cisco Unity Connection SIP integration

Voice mail Pilot



Voicemail Profile



SIP trunk to Unity Connection



Configuring SIP Trunk Security Profile to allow unsolicited NOTIFY for MWI

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/Cisco UCM/admin/7_1_2/ccmcfg/bccm-712-cm.html). You can also obtain the CISCO UCM SRND at: http://www.cisco.com/en/US/docs/voice_ip_comm/Cisco UCM/docguide/7_1_2/dg712.html




http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/bccm-712-cm.html

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

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 ! ! ! ! ! ! archive log config hidekeys ! ! ! ! ! ! interface FastEthernet0/0 ip address 192.X.X.X 255.255.255.0 duplex auto speed auto



! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 FastEthernet0/0 ! ! ip http server no ip http secure-server ! ! ! ! control-plane ! ! ! voice-port 0/0/0 timeouts ringing infinity ! voice-port 0/0/1 timeouts ringing infinity ! ! ! ! ! dial-peer voice 1999 voip description dial-peer for incoming and outgoing VoIP fax t38 calls destination-pattern 81.......... voice-class codec 1 session protocol sipv2 session target ipv4:192.X.X.Y �� incoming called-number 7323204075 dtmf-relay rtp-nte fax-relay sg3-to-g3 fax rate 14400 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none ! ! ! dial-peer voice 5068 pots destination-pattern 7323204075 port 0/0/1 forward-digits 0 ! line con 0 password cisco login line aux 0 line vty 0 4 exec-timeout 0 0 password cisco login ! scheduler allocate 20000 1000 end

This will be CISCO UCM’s IP address (SIP trunk)



c2801#

How to configure MGCP or H323 IOS gateway for fax T.38

Cisco IOS H323 fax t.38 configuration example: dial-peer voice 1999 voip description dial-peer for incoming and outgoing VoIP fax calls destination-pattern 81.......... voice-class codec 1 session protocol cisco session target ipv4:192.X.X.Y �� incoming called-number 7323204075 dtmf-relay rtp-nte fax-relay sg3-to-g3 fax rate 14400 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none ! ! dial-peer voice 5068 pots destination-pattern 7323204075 port 0/0/1 forward-digits 0

This will be CISCO UCM’s IP address (SIP trunk)



Cisco UCM H323 fax t.38 configuration example:



Cisco IOS MGCP fax t.38 configuration example:

ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server 192.X.X.X ccm-manager config ! mgcp mgcp call-agent 192.X.X.X 2427 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode nte-ca mgcp rtp unreachable timeout 1000 action notify mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package mgcp package-capability fm-package mgcp default-package fxr-package no mgcp package-capability res-package no mgcp timer receive-rtcp mgcp sdp simple mgcp fax rate 14400 mgcp fax t38 ls_redundancy 2 mgcp rtp payload-type g726r16 static ! mgcp profile default



Cisco UCM MGCP fax t.38 configuration example:



Configuring Fax over G711

Introduction: Cisco UCM 7.1(2) or older does not support mid-call codec change between two voice codecs, for example mid-call codec change between g729 to g711(It does support mid-call codec change between any voice codec to T.38 codec). The support for mid-call voice codec change will be supported in Cisco UCM 7.1(5) using SIP end-points only (SIP GW’s, SIP phones and SPI trunks). Because mid-call codec change is not supported by Cisco UCM, fax over a VoIP network not supporting T.38 must use a solution where fax calls can be routed out to the AT&T SIP trunk advertising G711 codec only. This can be achieved by providing to separate route patterns on Cisco UCM and two different outbound dial-peers on Cisco UBE, then using diferent steering digit to dial out the appropriate route-pattern/dial-peer combination. Also, on Cisco IOS gateways we recommend configuring the commands highlighted in the following configuration examples to optimize the rate of successful fax transmissions over a G711 call.

Cisco IOS MGCP fax over G711 configuration example:

ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server 192.X.X.X ccm-manager config ! mgcp mgcp call-agent 192.X.X.X 2427 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode nte-ca mgcp playout fixed 80 mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package mgcp package-capability fm-package mgcp default-package mt-package no mgcp timer receive-rtcp mgcp sdp simple mgcp fax rate 14400 mgcp fax t38 ecm mgcp fax t38 inhibit mgcp fax t38 ls_redundancy 2 ! mgcp profile default

Note: For MGCP gateways the playout delay command can only be set at a global level, therefore it is recommended that a separate MGCP gateway be used for voice (playout setting set at default (adaptive)) and a separate gateway be used for fax (playout delay setting set at fixed 80ms). If only one gateway is used it is up to the customer to decide how to configure the MGCP settings based on which application is considered highest priority and shall have optimized settings (fax having a higher success rate or voice having optimized transmission).



Cisco UCM MGCP fax over g711 configuration example:



Cisco IOS H323 fax over G711 configuration example:

dial-peer voice 19990 voip description incoming and outgoing dial-peer via CUCM to AT&T fax g711 destination-pattern 21[1-9][1-9][1-9]……. session target ipv4:192.168.201.254 incoming called-number [034][178][89][168] dtmf-relay rtp-nte playout-delay nominal 80 playout-delay mode fixed codec g711ulaw no vad ! ! dial-peer voice 5068 pots description pots dial-peer for incoming fax calls g711 destination-pattern [034][1780][896][1268] port 0/0/1 forward-digits 0



Cisco UCM H323 fax over g711 configuration example:



Cisco IOS SIP over G711 configuration example:

dial-peer voice 19990 voip description incoming and outgoing dial-peer via CUCM to AT&T fax g711 destination-pattern 21[1-9][1-9][1-9]……. session protocol sipv2 session target ipv4:192.168.201.254 incoming called-number [034][178][89][168] dtmf-relay rtp-nte playout-delay nominal 80 playout-delay mode fixed codec g711ulaw no vad ! ! dial-peer voice 5068 pots description pots dial-peer for incoming fax calls g711 destination-pattern [034][1780][896][1268] port 0/0/1 forward-digits 0



Cisco UCM SIP fax over g711 configuration example:



Configuring Cisco UCM second route-pattern and Cisco UBE second dial-peer to route outbound fax calls using fax over G711

This example shows an optional Route Pattern configured for G.711-only outbound calls. Typically, this would be used for fax/modem transmissions when G.711 codec is required for successful transmission. When fax/modem calls must be placed in this example a “2” must be dialed before the 10-digit telephone number, instead of the outside dial access code "8" used for voice calls. Furthermore, the "2" digit must not be dropped by the Cisco UCM pre-digit function, rather the digit "2" is sent to Cisco UBE to match the correct outbound g711-only dial-peer and the "2" is stripped by the Cisco UBE using a voice translation rule. Alternatively, a Route Pattern not requiring having to dial a different leading digit can be implemented. This Route Pattern is configured using a unique Route Partition, with parameter “Called Party Transformations Prefix Digit (Outgoing Calls)” configured with a prefix digit (using the CISCO UBE example configuration shown in the previous pages, this prefix would be “2”) . This partition is then assigned to a Calling Search Space that is assigned to fax machines/modems. When an outside telephone number is dialed using lines associated with this newly-created Calling Search Space, the Route Pattern assigned to this different partition is used in place of the standard outside dial access Route Pattern. Also, to ensure that inbound fax/modem calls are established using G.711, configure SIP trunks and/or MGCP/H.323 gateways supporting fax/modems into a Region using G.711 codec. Cisco UCM route pattern



Cisco UBE second dial-peer for g711-only calls

voice translation-rule 10 rule 1 /21/ /1\1/ ! voice translation-profile fax-call translate called 10 ! ! dial-peer voice 19995 voip description outgoing dial-peer to AT&T (fax only) translation-profile outgoing fax-call destination-pattern 21.......... session protocol sipv2 session target ipv4:207.Y.Y.Y incoming called-number 21.......... voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced dtmf-relay rtp-nte codec g711ulaw no vad FAX web link reference: http://www.cisco.com/en/US/docs/ios/voice/fax/configuration/guide/vf_cfg_t38_fxrly_ps6441_TSD_Products_Configuration_Guide_Chapter.html

http://www.cisco.com/en/US/docs/ios/voice/fax/configuration/guide/vf_cfg_t38_fxrly_ps6441_TSD_Products_Configuration_Guide_Chapter.html



Configuring Meetingplace Meetingplace version

Directory service admin

Note: This is the application user created in CISCO UCM



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 CISCO UCM 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)

http://tools.cisco.com/search/display?url=http%3A%2F%2Fwww.cisco.com%2Fen%2FUS%2Fdocs%2Fvoice_ip_comm%2Fmeetingplace%2F7x%2Fenglish%2Fbooks%2Fadmin_guides%2Fconfiguration_guide_7_0.html&pos=9&strqueryid=&websessionid=DKp6FmTY5EqiP8qFO4Yuocn

http://www.cisco.com/en/US/docs/voice_ip_comm/meetingplace/7x/english/books/admin_guides/configuration_guide_7_0.html



Configuring Cisco Unity Connection using SIP

General Settings



Telephony Integration



Phone System



Port Group



Edit server



Advanced Settings



Codec Advertising



Voicemail Ports



Adding a new voicemail port



Provisioning voicemail user



Setting or changing user vm password

Edit�Change Password



MWI settings



Note: To find detail configuration steps on how to integrate to Cisco Unity Connection using either SCCP or SIP go to: http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/roadmap/7xcucdg.html

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/roadmap/7xcucdg.html



AcronymsAcronymsAcronymsAcronyms

Acronym Definitions

SIP Session Initiation Protocol

MGCP Media Gateway Control Protocol

SCCP Skinny Client Control Protocol

CISCO UCM Cisco Unified Communications Manager

CISCO UBE 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 AAppendix AAppendix AAppendix 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 CISCO UBE IOS version , 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

http://wwwin-metrics.cisco.com/protected-cgi-bin/ddtsdisp.cgi?id=CSCsx62600



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

Cisco Unified Border Element web link references

DTMF feature support: http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-

dtmf_ps10591_TSD_Products_Configuration_Guide_Chapter.html

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

A

CUBE

AT&T IPFlexreach

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

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-dtmf_ps10591_TSD_Products_Configuration_Guide_Chapter.html

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-dtmf_ps10591_TSD_Products_Configuration_Guide_Chapter.html



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 version 0 ls-redundancy 0 hs-redundancy 0 fallback none fax-relay sg3-to-g3 Subscriber dial-peer voice 732321 voip description Outgoing dial-peer to Cisco Unified CM sub preference 126 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 version 0 ls-redundancy 0 hs-redundancy 0 fallback none fax-relay sg3-to-g3 ! ! Incoming call to CISCO UCM dial-peer dial-peer voice 732322 voip description incoming call to CISCO UCM 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 version 0 ls-redundancy 0 hs-redundancy 0 fallback none 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

26 The “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.



ImportImportImportImportant Informationant Informationant Informationant Information

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE

WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO

BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE

FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR

INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA

ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN

ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.



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