Cisco CallManager Express (CME) SIP Trunking

Configuration Example
Document ID: 91535

Contents
Introduction
Prerequisites
Requirements
Components Used
Conventions
SIP Protocol
CME SIP Trunk Support
DTMF Relay for SIP Trunks
Codec Support and Transcoding
Call Forward
Call Transfer
Call Hold
Configure
Network Diagram
Configurations
Verify
Troubleshoot
Troubleshooting Registration
Troubleshooting Call Setup
Related Information
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

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−bandSIP 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

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

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:
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 statusUse this command to display the status of E.164 numbers that a SIP
gateway has registered with an external primary SIP registrar.
• debug ccsip messageEnables 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 registeredVerifies ephone registration.

• show voip rtp connectionDisplays 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 callDisplays active UAC and user agent server (UAS) information on SIP calls.
• show call active voice briefDisplays active call information for voice calls or fax transmissions in
progress.

Debug commands:

• debug ccsip messageEnables all SIP SPI message tracing, such as those that are exchanged between
the SIP UAC and the access server.
• debug voip ccapi inoutTraces the execution path through the call control API.
• debug voice translationChecks 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−eventsEnables debugging for Real−Time Transport Protocol (RTP)
named events packets.
• debug sccp messageDisplays the sequence of the SCCP messages.

Related Information
• Cisco Unified Communications Manager Express System Administrator Guide
• Cisco Unity Express 2.3 Installation and Upgrade Guide
• Verifying and Troubleshooting SIP Features
• Managing and Monitoring Cisco Unified CallManager Express Systems
• Voice Technology Support
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Updated: Nov 16, 2007 Document ID: 91535