Elliot Murdock wrote: > [Jul 2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) --- > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 : > 5060 (no NAT) > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Using INVITE request as > basis request - 6998640000475636237-1246542986-18105 > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found no matching peer or > user for '216.48.184.50:5060' > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 18 > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 98 > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 101 > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 13 > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Capabilities: us - 0x8000e > (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing) > , combined - 0x0 (nothing)
And there it is... you have not allowed G.729 to be used by that peer in sip.conf. In addition, no peer or user in sip.conf was found to match the request, so unless you can correct that situation, you'll have to modify the allow/disallow options in the general section of sip.conf, since this call is being handled as an 'anonymous' peer. Asterisk properly parsed the SDP and understands that the peer supports G.729. None of the concerns about SDP parsing or RFC compliance, as it turns out, were even relevant to this problem :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
