Show us the sip debug for a failed call. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kamlesh Kumar Sent: Monday, May 27, 2013 2:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G.729 codec in pass-thru mode
Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456@default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456) -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0) -- <SIP/100-00000000>AGI Script call.php completed, returning 0 -- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL' If I use, ulaw, call works fine. Thanks, Kamlesh -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users