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<body class='hmmessage'><div dir='ltr'>Matthew,<BR> <BR>Yes that's correct, when I use u-law call works fine. <BR> <BR>In case of g729, I enabled sip debug with 'sip set debug on' and captured all the sip traces and got whatever I posted in last email. There was no other call on the system when I captured sip trace. Please suggest further proceedings.<BR> <BR>Regards,<BR>Kamlesh<br> <BR><div>> Date: Wed, 29 May 2013 08:42:39 -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 Kumar wrote:<br>> ><br>> > Call even doesn't go to the ITSP. I tried without AGI script and the results<br>> > were same.<br>> <br>> <br>> Kamlesh,<br>> <br>> Your first message stated that the call is successful if the codec is u-law, so<br>> there must be communication between the Asterisk server and the ITSP. The key<br>> to understanding why the G.729 call fails is in this SIP signaling.<br>> <br>> How are you capturing the SIP trace? Are you enabling SIP debugging for the<br>> specific SIP softphone? If so, please use "sip set debug on" to enable it for<br>> all SIP packets. Then wait until there are no other calls on the Asterisk<br>> server, try another G.729 call, and post the CLI output.<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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