Disconnected call on SIP trunk

[blive56133]

Hi.

We have 2 CUCM cluster connected via SIP trunk. When phone 1 calls phone 2 signalling should go through CUCM’s and RTP from phone 1-MTP-phone 2.

On SIP trunk I set up MRGL to use MTP.

Should Media Termination Point Required thick box be enabled?

When we do a test call, a call is established but there people on both sides cannot hear each other and the call gets disconnected after approximately 15-20 seconds.

Any idea what might be causing this?

[Rajan]

Hi,

MTP required checkbox needs to be checked in the SIP trunk configuration on both sides if you want MTP to be involved in the call.

Also make sure that you have only have the required MTP in the MRG/MRGL associated with the sip trunk.

HTH

Rajan

[blive56133]

When we enable this box and if I make a call, after I call the number I don't hear anything in my headset (no ringing tone and no rtp)

[Rajan]

If you have firewall in your network b/w these CUCM clusters, then you need to allow the IP address of MTP in that for RTP port numbers. Else you will have one way/no way audio issues.

Thanks,

Rajan

[blive56133]

Will check the firewall, thanks. But why does the call disconnects after 20seconds?

[Rajan]

Can you check what is the cause code we are getting for the disconnect in the CUCM traces sip messages. That will probably give an idea on this and whether there are any codec issues as well.