Hey Dan, and thanks a lot for your answer regarding Cisco CCM and MTP. I will continue my tests, and maybe give a try to the patch you mentionned. However, this will probably be too "cutting edge" for the project ;-) I have a few questions, though:
- You mention that Cisco indicates that any H323 trunk with advanced features needs an MTP. Can you point me to the place where you found this ? Because as far as I can tell, this is not true for a trunk to a Cisco gateway. - I have tested ooh323c from Asterisk-Addons. Reading what you wrote, I should have better luck with the Sourceforge version... - From your experience, do you feel that a clean CCM<->* integration is possible ? I am currently interested in simple feature (MoH, transfers, maybe Call Park). A friend of mine is working on the voicemail (unity) replacement/integration. Thanks again for you quick support, and sorry for my late answer ! BR, - Patrick - -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: vendredi, 21. octobre 2005 18:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? > Is it required to use an MTP on the Cisco callmanager, when integrating > with asterisk (using h323) ? As of CCM 4.X, Cisco indicates that any H.323 trunk that will support MoH/Transfer/etc need MTP resources. Annoying. > 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. Sure will drop the call. MTP does solve this. > The idea was to avoid RTP streams through the call manager. Good plan, and one that I would consider a must for scalability and quality. > 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. Odd, I am using ooh323c. I have a special test release, but the fixes for our CCM4 enviroment were added to CVS. Are you using ooh323c from Asterisk-Addons or a download from Open Systems? > 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 - http://bugs.digium.com/view.php?id=5374 has a patch that allows * to send RTP packets when it is not receiving them. I wasn't expecting this result, but applying this patch resolved the disconnect when a SCCP phone put a call on hold and allows transfers. The bug/patch got quite a bit of early attention, but seems to have languished. Try it out and provide feedback. Maybe enough success reports will help get it rolling again. Dan _______________________________________________ --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
