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<body class='hmmessage'><div dir='ltr'>Hello Matthew,<BR> <BR>Call even doesn't go to the ITSP. I tried without AGI script and the results were same.<BR> <BR>Regards,<BR>Kamlesh<br> <BR><div>> Date: Tue, 28 May 2013 18:32:19 -0500<br>> From: mroth@imminc.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode<br>> <br>> Kamlesh,<br>> <br>> Please provide SIP traces of both call legs for a failed call.<br>> <br>> Your last message only included a SIP trace of the call leg from the SIP<br>> softphone to the Asterisk server. There was no SIP trace for the call leg from<br>> the Asterisk server to the ITSP and, as shown below, that is probably where the<br>> answer to your problem can be found.<br>> <br>> First, the call leg from the SIP softphone to the Asterisk server successfully<br>> negotiated G.729 as the codec:<br>> <br>> > [May 28 11:51:34] Found RTP audio format 18<br>> > ...<br>> > [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)<br>> <br>> However, the "call.php" AGI script then failed to create the call leg from the<br>> Asterisk server to the ITSP:<br>> <br>> > [May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "call.php")<br>> > [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php<br>> > [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)<br>> > [May 28 11:51:34] == Using SIP RTP CoS mark 5<br>> > [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456<br>> > [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071ad6@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)<br>> > [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0)<br>> > [May 28 11:51:34] -- <SIP/100-0000115f>AGI Script call.php completed, returning 0<br>> > [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-0000115f' status is 'CHANUNAVAIL'<br>> <br>> Regards,<br>> <br>> Matthew Roth<br>> InterMedia Marketing Solutions<br>> Software Engineer and Systems Developer<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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