Call recording on Cisco Call Manager

One of the big feature that Cisco CallManager don’t have into the native setup is CALL RECORDING.

Based on this article https://www.ucguru.com/recording-call-manager-calls-asterisk/ and our knowledge on Cisco UC and Asterisk we have created a READY-TO-USE virtual appliance to enable Cisco’s phone to be recorded on LAN.

With this setup every recorded phones create a new call-leg and send a copy of RTP stream to our Asterisk box using a standard Trunk SIP.

Call can be recorded indistinctly with a always-record function or with on-demand recording using the softkey on the phones.

Note: on-demand recording is supported only in CallManager version > 9.0

Phones must have a built-in-bridge support enable to record calls, like these models:

7906G, 7911G, 7931G, 7941G, 7941G-GE, 7961G, 7961G-GE, 7970G, 7971G-GE,7942G,7962G,7945G,7965G, 7975G and latest models.

Before deploy the recording appliance must enable builtin-bridge feature on CUCM:

System -> Service parameters and select Cisco CallManager service

Turn Builtin Bridge => ON

Increase SIP Expire Timers

Cisco Call Manager and Asterisk recorder connect each-other with a SIP Trunk. Bellow my configuration:

In my case the Asterisk box have IP 192.168.199.23 and we use the default SIP port 5060

Check to have a security profile for the SIP trunk with the following settings:

Now must create a new recording profile with a DN/Pilot used to reach the Asterisk recorder. In our case 1111200

Select the default CSS for the AstRecorder profile.

Create a route pattern to forward number 1111200 to the new SIP trunk of Asterisk Recorder.

If on-demand recording is needed must add to the default softkey template (or the group with recording enable) the RECORD button on CONNECTED state.

Now for every DNs / Phones recorded must enable these two settings:

Asterisk recorder is based on Elastix distro and the default settings are:

username: root/admin

password: password

default IP: 192.168.99.23

After importing the VM into your Vmware infrastructure and change the IP to fit your network (from console or web interface through System -> Network menu) you need to set the IP of your Cisco CUCM into the SIP trunk. In my case 192.168.199.2

Change the internal network and NAT configuration (if needed) into:

Recorder calls can be listen through Web Interface or downloaded from the SMB Share.

TROUBLESHOOTING

On the web page of Cisco phone check the second stream during calls, must point to Asterisk recorder.