Dinesh Nair wrote:
the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up.
I had a similar problem connecting Asterisk to an Avaya IP403 via OOH323: In the end, I removed all the disallow=all and allow=<codec> lines in Asterisk. This seems to have allowed the two systems to overcome the codec negotiation problems they were having and proceed with actual audio transfer. :)
I have no idea if this is related, but I thought I'd just throw that out there, if only for testing purposes.
cYa, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .....>> Open Source - Own it - Squiz.net ...../> _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
