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 - _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
