Matthew, Yes that's correct, when I use u-law call works fine. 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. Regards, Kamlesh > Date: Wed, 29 May 2013 08:42:39 -0500 > From: mr...@imminc.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] G.729 codec in pass-thru mode > > Kamlesh Kumar wrote: > > > > Call even doesn't go to the ITSP. I tried without AGI script and the results > > were same. > > > Kamlesh, > > Your first message stated that the call is successful if the codec is u-law, > so > there must be communication between the Asterisk server and the ITSP. The key > to understanding why the G.729 call fails is in this SIP signaling. > > How are you capturing the SIP trace? Are you enabling SIP debugging for the > specific SIP softphone? If so, please use "sip set debug on" to enable it for > all SIP packets. Then wait until there are no other calls on the Asterisk > server, try another G.729 call, and post the CLI output. > > 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
