I had it 'working' and quickly moved back to H.323 using chan_oh323. I had great expectations for SIP support in CCM4, but ended up disappointed. The requirement for a Media Termination Point to handle DTMF and support for only the G.711 codecs was just too much.
I'm hoping that Cisco will continue to improve the CCM SIP trunk and it will become a viable option, but at the moment it is really only usefull as something to experiment with. Dan -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Parker, Blake (MIS) Sent: Wednesday, March 23, 2005 5:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] * and Cisco Callmanager Interconnection Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake _______________________________________________ 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 _______________________________________________ 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
