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Cisco Unified Communications Manager Express

Cisco CallManager Express (CME) SIP Trunking Configuration Example

Document ID: 91535

Updated: Nov 16, 2007

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Introduction

Today, the telecommunications industry is in the process of making the transition from long establishing switching and transport techonologies to IP-based transport and edge devices. The IP communication revolution has started to create a tremendous commercial impact in small and medium businesses. These small and medium businesses are realizing that the use of IP is very efficient because IP can use Voice, Video, and Data capabilities over a single network, instead of using three separate special-purpose networks. Figure 1 shows an IP telephony deployment trending towards IP trunking.

Figure 1 - IP Telephony System

cme-sip-trunking-config1.gif

IP PBXs are starting to predominate in the business of the Voice technology, and the TDM PBXs are no longer the primary source as the crossover going between two Voice networks. The usage of the TDM PBXs has decreased in the last couple of years, and the use of the IP PBX is becoming a good investment in IP LANs and WANs. In order to connect to the PSTN, PBXs need some sort of trunking such as TDM (T1/E1) or analog lines. IP PBXs can access the PSTN using these types of trunks, but need a media gateway that converts the IP voice traffic to traditional PSTN, which sometimes can result in successive translation from IP domain to TDM domain. These successive translations increase the maintenance costs of the gateways, increases latency, and reduces voice quality.

In order to avoid these problems, the IP PBXs use protocols for session initiation and management, the most prominent of which is Session Initiation Protocol (SIP). This document provides a description on SIP trunking and Cisco CallManager Express (CME), and a configuration to implement an IP-based telephony system with CME using SIP trunking for inbound and outbound calls.

Prerequisites

Requirements

Ensure that you meet these requirements before you attempt this configuration:

  • CME release 4.1 is installed

  • An image of Cisco IOS® Software Release 12.4(11)XJ or IOS 12.4(6th)T is on the router

  • An NM-CUE module is installed with CUE release 2.3.4

Components Used

The information in this document is based on these software and hardware versions:

  • Cisco 3825 Router on Cisco IOS Software Release 12.4(11)XJ

  • Cisco Catalyst 3550 Switch on Cisco IOS Software Release 12.4

  • Cisco IP 7960 Phone

  • Cisco CallManager Express 4.1

  • Cisco Unity Express 2.3.4

The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.

Conventions

Refer to the Cisco Technical Tips Conventions for more information on document conventions.

SIP Protocol

SIP is an ASCII based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints. SIP has rapidly emerged as the standard protocol used in IP communications, because it is a multimedia protocol that can be used for video sessions and instant messaging in addition to voice. Also, SIP can handle conference sessions and broadcasts, as well as one-to-one sessions. SIP has great potential in transforming and developing the way people communicate. For this reason, Cisco has and continues to play an important role in taking a leadership to create new technologies that make SIP and its applications the standard of IP communications.

SIP trunks are similar to a phone line, except that SIP trunks use the IP network, not the PSTN. In addition, SIP trunks permit the convergence of voice and data onto common all-IP connections. In order to access the IP network using an SIP trunk, it is necessary that configurations be made on the service provider, as well as on the customer side. Customers need to set and configure CME, which is the PBX that will interpret the SIP signal adequately and pass traffic successfully. The service provider needs to configure an SIP Proxy Server. However, SIP trunks are more complicated to establish than regular PSTN trunks. The reason is that a customer faces challenges in handling different interpretation and implementations of SIP by equipment vendors, delivering security, managing quality of service (QoS), enabling Network Address Translation (NAT) and firewall traversal, and ensuring carrier-grade reliability and continuity of service.

These points describe why SIP trunks are becoming so apparent in small and medium businesses:

  • Quick and Easy Deployment

  • Improved Utilization of Network Capacity

  • Potential for Consolidating and Lowering Telephony Costs

  • Economical Direct Inward Dial (DID)

  • Business Continuity

CME SIP Trunk Support

Cisco CME is an IP telephony solution that is integrated directly into Cisco IOS software. CME permits small and medium businesses to deploy voice, data, and video on a single platform. An IP telephony network is simple to set because CME runs on a single router, which delivers a PBX functionality for businesses. Therefore, by using CME, small and medium businesses can deliver IP telephony and data routing using a single converged solution with minimal costs.

DTMF Relay for SIP Trunks

CME started to support SIP trunking when CME 3.1 was released. However, some problems existed when an SIP phone called an SCCP phone or tried to access voicemail. The problem is that SCCP phones connected to CME require the use of out-of-band DTMF relay to transport DTMF (digits) across VoIP connections, and SIP phones use in-band tranports. A DTMF distortion existed between the two devices. When CME 3.2 was released, support was added to the DTMF relay. DTMF digits from SCCP could be converted to in-band DTMF relay mechanism through RFC2833 or Notify methods.

CME currently supports this list of DTMF internetworking for SIP to SIP calls:

  • Notify <---> Notify since 12.4(4)T

  • RFC2833 <---> Notify since 12.4(4)T

  • Notify <---> RFC2833 since 12.4(4)T

  • Inband G711 <---> since 12.4(11)T [Requires Transcoder]

CME currently supports this DTMF internetworking for SIP to SCCP calls:

  • SCCP out-of-band—SIP Notify / RFC2833 since 12.4(4)T

Codec Support and Transcoding

Another important aspect to consider when you set up an SIP trunk is the codecs supported. Codecs represent the pulse-code modulation sample for signals in voice frequencies. SIP trunks support these codecs: G.711 and G.729. However, for different features such as Cisco Unity Express (CUE) and Music on Hold (MOH), only codec G.711 is supported. This means that voice calls that use SIP trunks using codec G.729 cannot access CUE, unless a transcoder exists to permit the compression and decompression of voice streams to match the CUE capabilities. MOH can also use codec G.729 to save bandwidth, but the codec does not provide adequate quality MOH streams. This is due to the fact that G.729 is optimized for speech. Therefore, you must force MOH to use G.711.

Call Forward

When a call comes in on an SIP trunk and gets forwarded (CFNA / CFB / CFA), then the default behavior is for the CME to send the 302 "Moved Temporarily" SIP message to the Service Provider (SP) proxy. The user portion of the Contact Header in the 302 message might need to be translated to reflect a DID that the SP proxy can route to. The host portion of the Contact Header in the 302 message should be modified to reflect the Address of Record (AOR) using the host-registrar CLI under sip-ua and the b2bua CLI under the VoIP dial peer going to the CUE.

Some SIP proxies might not support this. If so, then you need to add this:

Router(config)#voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

Figure 2 shows the behavior of the CME system when the 302 message is disabled.

Figure 2 - Call Forward Busy (CFB) flow with 302 message disabled

cme-sip-trunking-config2.gif

This method will allow hairpinning of the 302 SIP messages for call forwards on the CME. The above is also required if there are certain extensions that have no DID mapping as the SP proxy might not know how to route such calls. If you disable the 3xx response, the calling-number initiator can be used to preserve the caller ID of the original calling party.

Call Transfer

When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:

Router(config)#voice service voip
Router(conf-voi-serv)#no supplementary-service sip refer

Figure 3 shows the behavior of the CME system with the REFER method disabled.

Figure 3 - Transfer with REFER disabled

cme-sip-trunking-config3.gif

If REFER is supported on the SIP proxy, the user portion of the Refer-To and Referred-By must be translated to a DID that the SP proxy understands. The host portion of the Refer-To and Referred-By fields must be an IP address or DNS that the SP proxy can route to as well (this occurs by default on CME 4.1).

Call Hold

If an SCCP phone places a call from PSTN on HOLD, the CME locally changes the media. No SIP messages are sent across on the SIP trunk. Music on Hold will be played to the user across the SIP trunk based on the CME configuration.

Configure

In this section, you are presented with the information to configure the features described in this document.

Note: Use the Command Lookup Tool (registered customers only) to obtain more information on the commands used in this section.

Network Diagram

This document uses this network setup:

cme-sip-trunking-config4.gif

Configurations

These configuration elements provide an outline of the steps required to configure your CME with SIP trunks:

  • Infrastructure Elements: Interfaces, TFTP and DHCP services, NTP, etc

  • Telephony-service: Enables IOS "PBX" call control on the CME platform including elements of phone management

  • Ephones an Ephones-dns: Define IP phones and their telephone numbers

  • Dial Plan: Dial-peers, extensions, voice-translation rules

  • IOS SIP Configuration: Enables SIP, phone registration with SIP proxy, call routing over trunks, etc

  • Voicemail Support: Cisco Unity Express

  • Switch Catalyst Configuration: IP address, Interfaces, etc

This is the complete configuration needed to deploy a CME system with SIP trunks:

Router - CME Configuration
!
AUSNML-3825-01#show run
Building configuration...

Current configuration : 8634 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname AUSNML-3825-01
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$vBU1$MCMG1rXM5ejME8Wap6W0H1
!
no aaa new-model
clock timezone central -8
clock summer-time central recurring
ip cef
!

!--- DHCP Configuration ---

ip dhcp pool Voice
   network 172.22.100.0 255.255.255.0
   option 150 ip 172.22.1.107 
   default-router 172.22.100.1 
!
ip dhcp pool Data
   network 172.22.101.0 255.255.255.0
   option 150 ip 172.22.1.107 
   default-router 172.22.101.1 
!
!
ip domain name cisco.com
ip name-server 205.152.0.20
multilink bundle-name authenticated
!
voice-card 0
 no dspfarm
!
!
!
!

!--- Voice Class and Service VoIP Configuration ---

voice service voip 
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily

!---Disable 302 sending

 no supplementary-service sip refer

!---Disable REFER sending

 sip
  registrar server expires max 3600 min 3600
  localhost dns:domain.test.com 
!
!
voice class codec 1
 codec preference 1 g711ulaw
!
!
!
!
!
!
!
!
!
!
!

!--- Voice Translation Rules ---

voice translation-rule 1
 rule 1 /5123781291/ /601/

!--- An inbound rule for AA pilot "601

 rule 2 /5123781290/ /600/

!--- An inbound rule for the voicemail pilot "600"

!
voice translation-rule 2
 rule 1 /^911$/ /911/

!--- An outbound rule to allow "911"

 rule 2 /^9\(.*\)/ /\1/


!--- An outbound rule to strip "9" from PSTN calls

!
voice translation-rule 3
 rule 1 /^.*/ /5123781291/

!--- An outbound rule to change calling-number CLID to a
!--- "main" number

!
voice translation-rule 4
 rule 1 /^9(.......)$/ /512\1/

!--- An outbound rule to add areacode for local calls

 rule 2 /600/ /5123788000/

!--- An outbound rule to present the voicemail pilot extension as DID

 rule 3 /601/ /5123788001/

!--- An outbound rule to present the AA pilot extension as DID

 rule 4 /^2(..)$/ /51237812\1/

!--- An outbound rule to support transfers and call-forwards

 rule 5 /^9(.*)/ /\1/

!--- An outbound rule to strip "9" from "9+" transfers and call-forwards

!
!
voice translation-profile CUE_Voicemail/AutoAttendant

!--- Applied to the inbound dial-peers for CUE

 translate called 1
!
voice translation-profile PSTN_CallForwarding

!--- Applied to CUE dial-peers

 translate redirect-target 4
 translate redirect-called 4
!
voice translation-profile PSTN_Outgoing

!--- Applied to all outbound dial-peers

 translate calling 3
 translate called 2
 translate redirect-target 4
 translate redirect-called 4
!
!
!
!
!
!
!
vlan internal allocation policy ascending
!
!
!
!

!--- Internet Connection Configuration ---

interface GigabitEthernet0/0
 no ip address
 duplex auto
 speed auto
 media-type rj45
 no keepalive
!
interface GigabitEthernet0/0.1
 encapsulation dot1Q 1 native
 ip address 172.22.1.71 255.255.255.0
!
interface GigabitEthernet0/0.20
 encapsulation dot1Q 20
 ip address 172.22.101.1 255.255.255.0
!
interface GigabitEthernet0/0.100
 encapsulation dot1Q 100
 ip address 172.22.100.1 255.255.255.0
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
 no keepalive
!
interface Service-Engine1/0
 ip unnumbered GigabitEthernet0/0.1
 service-module ip address 172.22.1.253 255.255.255.0
 service-module ip default-gateway 172.22.1.71
!
ip route 0.0.0.0 0.0.0.0 172.22.1.1
ip route 172.22.1.253 255.255.255.255 Service-Engine1/0
!
!
ip http server
no ip http secure-server
!
!
!

!--- TFTP Server Configuration  ---

tftp-server flash:P0030702T023.bin
tftp-server flash:P0030702T023.loads
tftp-server flash:P0030702T023.sb2
tftp-server flash:P0030702T023.sbn
!
control-plane
!
!
!
!
!
!
!

!--- SIP Trunk Configuration ---

dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 translation-profile incoming CUE_Voicemail/AutoAttendant
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 2 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9........
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 3 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9[2-9]..[2-9]......
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 4 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9[0-1][2-9]..[2-9]......
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 5 voip
 description **911 Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 911
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 6 voip
 description **Emergency Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9911
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 7 voip
 description **911/411 Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9[2-9]11
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 8 voip
 description **International Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9011T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 9 voip
 description **Star Code to SIP Trunk**
 destination-pattern *..
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!

!--- Voicemail Configuration ---

dial-peer voice 10 voip
 description **CUE Voicemail**
 translation-profile outgoing PSTN_CallForwarding
 destination-pattern 600
 b2bua

!--- Used by CME to send its IP address to SP proxy instead of CUE

 session protocol sipv2
 session target ipv4:172.22.1.155
 dtmf-relay sip-notify

!--- This can also be RFC2833 going to CUE

 codec g711ulaw

!--- CUE only supports G711ulaw as the codec

 no vad

!--- With VAD enabled, messages left on CUE could be blank or poor quality

!
!
!
dial-peer voice 11 voip
 description **CUE Auto Attendant**
 translation-profile outgoing PSTN_CallForwarding
 destination-pattern 601
 b2bua
 session protocol sipv2
 session target ipv4:172.22.1.155
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
!

!--- SIP UA Configuration ---

sip-ua 
 authentication username 5123781000 password 075A701E1D5E415447425B
 no remote-party-id
 retry invite 2
 retry register 10
 retry options 0
 timers connect 100
 registrar dns:domain.test.com expires 3600
 sip-server dns:domain.test.com
  host-registrar
!
!

!--- CME Telephony Service Configuration ---

telephony-service
 no auto-reg-ephone
 load 7960-7940 P0030702T023
 max-ephones 168
 max-dn 500
 ip source-address 172.22.1.107 port 2000
 calling-number initiator

!--- Preserves the caller-id of a call when transferred or forwarded

 dialplan-pattern 1 51237812.. extension-length 3 extension-pattern 2.. no-reg
 voicemail 600
 max-conferences 12 gain -6
 call-forward pattern .T
 call-forward system redirecting-expanded

!--- Enables translation rule features for call-forwarding

 moh music-on-hold.au
 transfer-system full-consult dss
 transfer-pattern 9.T
 secondary-dialtone 9
 create cnf-files version-stamp Jan 01 2002 00:00:00
!
!

!--- Ephone and Ephone-dn Configuration ---

ephone-dn  11  dual-line
 number 201 secondary 5123781201 no-reg both

!---"no-reg both" means do not try to register either extension with SP SIP Proxy

 name John Smith
 call-forward busy 600
 call-forward noan 600 timeout 15
!
!
ephone-dn  12  dual-line
 number 202 secondary 5123781202 no-reg both
 name Enrique Zurita
 call-forward busy 600
 call-forward noan 600 timeout 15
!
!
ephone-dn  13
 number 5123788000
 description **DID Number for Voicemail**
!
!
ephone-dn  14
 number 5123788001
 description **DID Number for Auto Attendant*
!
!
ephone-dn  15
 number 8000... no-reg primary
 mwi on
!
!
ephone-dn  16
 number 8001... no-reg primary
 mwi off
!
!
ephone  1
 mac-address 0008.A371.28E9
 type 7960
 button  1:11
!
!
!
ephone  2
 mac-address 0008.A346.5C7F
 type 7960
 button  1:12
!
!
!
!
line con 0
 stopbits 1
line aux 0
 stopbits 1
line 66
 no activation-character
 no exec
 transport preferred none
 transport input all
 transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
 password ut69coe
 login
!
scheduler allocate 20000 1000
ntp server 172.22.1.107
!
end

Router - CUE Configuration
se-172-22-1-253#show run
 
Generating configuration:


clock timezone America/Chicago

hostname se-172-22-1-253

ip domain-name localdomain



groupname Administrators create
groupname Broadcasters create


!--- Users ---

username Enrique create
username John create
username Enrique phonenumberE164 "5123781202"
username John phonenumberE164 "5123781201"
username Enrique phonenumber "202"
username John phonenumber "201"


!--- AutoAttendant ---

ccn application autoattendant
 description "**AutoAttendant**"
 enabled
 maxsessions 4
 script "aa.aef"
 parameter "busOpenPrompt" "AABusinessOpen.wav"
 parameter "operExtn" "601"
 parameter "welcomePrompt" "AAWelcome.wav"
 parameter "disconnectAfterMenu" "false"
 parameter "busClosedPrompt" "AABusinessClosed.wav"
 parameter "allowExternalTransfers" "false"
 parameter "holidayPrompt" "AAHolidayPrompt.wav"
 parameter "businessSchedule" "systemschedule"
 parameter "MaxRetry" "3"
 end application


!--- MWI ---

ccn application ciscomwiapplication
 description "ciscomwiapplication"
 enabled
 maxsessions 8
 script "setmwi.aef"
 parameter "CallControlGroupID" "0"
 parameter "strMWI_OFF_DN" "8001"
 parameter "strMWI_ON_DN" "8000"
 end application


!--- Voicemail ---

ccn application voicemail
 description "**Voicemail**"
 enabled
 maxsessions 4
 script "voicebrowser.aef"
 parameter "uri" "http://localhost/voicemail/vxmlscripts/login.vxml"
 parameter "logoutUri" "http://localhost/voicemail/vxmlscripts/mbxLogout.jsp"
 end application


!--- SIP ---

ccn subsystem sip
 gateway address "172.22.100.1"

!--- Must match the "ip source-address" in telephony-service

 dtmf-relay sip-notify
 mwi sip outcall

!--- Subscribe / Notify and Unsolicited Notify have not been tested

 transfer-mode blind bye-also

!--- Testing with REFER method on CUE has caused certain call flows to break

 end subsystem


!--- Trigger Phones ---

ccn trigger sip phonenumber 600
 application "voicemail"
 enabled
 maxsessions 4
 end trigger

ccn trigger sip phonenumber 601
 application "autoattendant"
 enabled
 maxsessions 4
 end trigger

service phone-authentication
 end phone-authentication

service voiceview
 enable
 end voiceview


!--- Voicemail Mailboxes ---

voicemail default mailboxsize 21120
voicemail broadcast recording time 300

voicemail mailbox owner "Enrique" size 300
 description "**Enrique_Mailbox**"
 expiration time 10
 messagesize 120
 end mailbox

voicemail mailbox owner "John" size 300
 description "**John'sMailbox**"
 expiration time 10
 messagesize 120
 end mailbox

end

Switch Configuration

!--- Interface Connected to CME/CUE Router ---

interface FastEthernet0/2
 description Trunk to 3825
 switchport trunk encapsulation dot1q
 switchport mode trunk
 no ip address
 duplex full
 speed 100


!--- Interfaces Connected to the IP Phones ---

interface FastEthernet0/7
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 20

!--- Data Traffic ---

 switchport mode trunk
 switchport voice vlan 100

!--- Voice Traffic ---

 no ip address
 spanning-tree portfast


interface FastEthernet0/8
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 20
 switchport mode trunk
 switchport voice vlan 100
 no ip address
 spanning-tree portfast
 

!--- IP Address ---

interface Vlan1
 ip address 172.22.1.194 255.255.255.0
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.22.1.1
ip http server

Verify

There is currently no verification procedure available for this configuration.

Troubleshoot

This section provides information you can use to troubleshoot your configuration.

The Output Interpreter Tool (registered customers only) (OIT) supports certain show commands. Use the OIT to view an analysis of show command output.

Note: Refer to Important Information on Debug Commands before you use debug commands.

Troubleshooting Registration

Troubleshooting the SIP trunk on CME involves the same commands you use for IOS SIP GW troubleshooting and CME troubleshooting. Use these commands in order to check if your DN is registered:

  • show sip-ua register status—Use this command to display the status of E.164 numbers that a SIP gateway has registered with an external primary SIP registrar.

  • debug ccsip message—Enables all SIP SPI message tracing, such as those that are exchanged between the SIP user-agent client (UAC) and the access server.

Troubleshooting Call Setup

Commands for troubleshooting calls over SIP trunks are essentially the same as you use for regular SIP GW and CME troubleshooting.

Show commands:

  • show ephone registered—Verifies ephone registration.

  • show voip rtp connection—Displays information about RTP named-event packets, such as caller-ID number, IP address, and ports for both the local and remote endpoints.

  • show sip-ua call—Displays active UAC and user agent server (UAS) information on SIP calls.

  • show call active voice brief—Displays active call information for voice calls or fax transmissions in progress.

Debug commands:

  • debug ccsip message—Enables all SIP SPI message tracing, such as those that are exchanged between the SIP UAC and the access server.

  • debug voip ccapi inout—Traces the execution path through the call control API.

  • debug voice translation—Checks the functionality of a translation rule.

  • debug ephone detail mac-address <mac of phone> —Sets detail debugging for the Cisco IP phone.

  • debug voip rtp session named-events—Enables debugging for Real-Time Transport Protocol (RTP) named events packets.

  • debug sccp message—Displays the sequence of the SCCP messages.

Related Information

Updated: Nov 16, 2007
Document ID: 91535