If using CCM >= 4.0, using SIP trunks will alleviate a lot of headaches.
On Tue, 2005-11-15 at 16:33 -0800, Dan Austin wrote: > I posted a couple weeks back about our experiences with H323 trunks on > CCM. > As of version 4.0, the Cisco documents state that a 3rd party H323 > gateway > requires a Media Termination Point.. > > At the time I said that I have Asterisk working with the ooH323c > version of > chan_h323 with out an MTP. I just found that another engineer had > been > twiddling with the CCM config, and we were using a MTP. > > I retested chan_h323 without the MTP, and indeed per the Cisco docs, > when a phone connected to CCM puts a call placed through chan_h323 on > hold, the call is disconnected. This IS NOT a bug with asterisk or > the > chan_h323, but a known Cisco quirk. > > Cisco's own H323 gateways are capable of dynamically > creating/connecting > to a MTP. Which permits calls to/through them to allow rtp re-invites > and > still preserve a call during media transitions. > > I thought I should post this for the archives in case anyone searching > for > details about connecting CCM to Asterisk found my earlier > misinformation. > > 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