Thanks for this one, Greg ! -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver Sent: mardi, 1. novembre 2005 16:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ?
You will probably also need to change the media exchange timers in CCM if you are going to use it as a PRI gateway - otherwise asterisk -> 323 -> CCM -> PSTN calls will get dropped after 4 secs of ringing. On Mon, 2005-10-31 at 14:41 +0100, Patrick Zwahlen wrote: > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users