Configure a SIP Trunk between Cisco CME and Cisco CUCM

Introduction :

In this post, I describe a basic configuration of SIP Trunk between Cisco CME (v4.1) and Cisco CUCM (v8.6.2).
Before this, if you want to know how to add ephone and ephone-dn in CME follow this post :

Basic Ciso CME Configuration – Place a simple call

Schema :


Cisco CME Configuration :

To configure a SIP Trunk, we need to configure two dial-peers one for incoming call and the other one for outgoing call.

In the dial peer for outgoing call, we set the destination pattern which specifies the route pattern to reach our phones on CUCM, (so 1…)

dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 session protocol sipv2
 session target sip-server
dial-peer voice 2 voip
 description **Outgoing Call to SIP Trunk**
 destination-pattern 1...
 session protocol sipv2
 session target sip-server
 sip-server ipv4:

Cisco CUCM Configuration :

SIP Trunk Security profile with UDP :

By default the CME SIP Trunk is configured to use UDP as transport protocol, so we need to configure a SIP Trunk security profile with UDP on CUCM.

Go to System > Security > SIP Trunk Security Profile


SIP Trunk Configuration :

Go to Device > Trunk

Use default configuration for all parameters. If your DN are placed in the None Partition, you won’t need to configure CSS otherwise you need to specify the good CSS to reach your phones.

For SIP Trunk Security Profile, use the previous created “Non Secure UDP SIP Trunk Profile”.

For destination enter the IP of your CME (in my lab as destination address.

Test call :
With this configuration. The signaling is OK but media not.

With a wireshark trace, we can see by default G729 codec is used. Also, CME stays in the media path.

Usefull show command :

R2#show dial-peer voice summary
dial-peer hunt 0
             AD                                    PRE PASS                OUT
20001  pots  up   up             2000$              0                           50/0/1
20002  pots  up   up             2001$              0                           50/0/2
1      voip  up   down                              0  syst sip-server
2      voip  up   up             1...               0  syst sip-server

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