Hello Matthew, Call even doesn't go to the ITSP. I tried without AGI script and the results were same. Regards, Kamlesh > Date: Tue, 28 May 2013 18:32:19 -0500 > From: [email protected] > To: [email protected] > Subject: Re: [asterisk-users] G.729 codec in pass-thru mode > > Kamlesh, > > Please provide SIP traces of both call legs for a failed call. > > Your last message only included a SIP trace of the call leg from the SIP > softphone to the Asterisk server. There was no SIP trace for the call leg > from > the Asterisk server to the ITSP and, as shown below, that is probably where > the > answer to your problem can be found. > > First, the call leg from the SIP softphone to the Asterisk server successfully > negotiated G.729 as the codec: > > > [May 28 11:51:34] Found RTP audio format 18 > > ... > > [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 > > (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) > > However, the "call.php" AGI script then failed to create the call leg from the > Asterisk server to the ITSP: > > > [May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "call.php") > > [May 28 11:51:34] -- Launched AGI Script > > /var/lib/asterisk/agi-bin/call.php > > [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: > > (SIP/yyy.yyy.yyy.yyy/12127773456) > > [May 28 11:51:34] == Using SIP RTP CoS mark 5 > > [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 > > [May 28 11:51:34] Scheduling destruction of SIP dialog > > '[email protected]' in 32000 ms (Method: > > INVITE) > > [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) > > [May 28 11:51:34] -- <SIP/100-0000115f>AGI Script call.php completed, > > returning 0 > > [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-0000115f' > > status is 'CHANUNAVAIL' > > Regards, > > Matthew Roth > InterMedia Marketing Solutions > Software Engineer and Systems Developer > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
