[Asterisk-Users] MTP required for CCM integration ?

[Patrick Zwahlen]

Hi,

Is it required to use an MTP on the Cisco callmanager, when integrating
with asterisk (using h323) ?

I am working on a project where the goal is to interconnect Cisco
Callmanager (version 4) clouds together, using either SIP or IAX between
multiple * servers. Basic setup will be:

PHONE - sccp - CCM (V4) - h323 - ASTERISK - iax - ASTERISK - h323 - CCM
- sccp - PHONE

I am working on the first half of it, which is:

7920 --- SCCP --- CALLMANAGER (V4) --- chan_oh323 --- ASTERISK 1.0.9

I want to avoid the use of a gatekeeper.

In that configuration, I am trying to get call transfer working. The
phone can call the DEMO app on asterisk, but then I cannot transfer the
call to another Cisco phone (on the same callmanager). I have some PCAP
traces if required. Basically, the 2nd phone rings, but there is no
audio channel. After about 10 seconds, I see that that chan_oh323 hangs
up the call.

The idea was to avoid RTP streams through the call manager.

Now, if I define a Media Termination Point (MTP) on the Callmanager,
things work much better.

I have also tried the new ooh323 with 1.2.0-beta1, but I couldn't get
audio at all.

I have read a lot about people having success with integratin CCM and *,
but without any details, especially around MTP configuration.

Any help would be greatly appreciated. BR, - Patrick -