SIP_CODEC is only useable on a SIP channel. You can specify DAHDI codecs in users.conf.
From: [email protected] [mailto:[email protected]] On Behalf Of Ali Pey Sent: Friday, November 02, 2012 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Different codec for different type of calls Qasim, Thank you for your response. I tried it but still doesn't work. This is what I have: exten => _XXX.,1,NoOP(Set G711 codec) exten => _XXX.,n,Set(SIP_CODEC=ulaw) exten => _XXX.,n,Set(SIP_CODEC_OUTBOUND=ulaw) exten => _XXX.,n,Dial(DAHDI/g1/$EXTEN) Then I get this error: WARNING[12156]: channel.c:5796 ast_request: No translator path exists for channel type DAHDI (native (ulaw|alaw|slin)) to (h264|silk8) WARNING[12156]: app_dial.c:2277 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) I tried both g711 and ulaw with no luck. I read the SIP_CODEC value and it is set properly. Any suggestions/ideas? Thanks, Ali Pey On Thu, Nov 1, 2012 at 12:02 PM, [email protected] <[email protected]> wrote: exten => _X.,1,NoOP(G711 CoDec) exten => _X.,n,Set(SIP_CODEC=g711) exten => _X.,n,Dial(...) ${SIP_CODEC}: Set the SIP codec for the inbound (=first) call leg (see channelvariables.txt or README.variables in 1.2); Asterisk 1.6.2 also comes with SIP_CODEC_OUTBOUND <https://issues.asterisk.org/view.php?id=13243> for the remote (=second) call leg. Hope this helps, Regards, Qasim On Thu, Nov 1, 2012 at 8:49 PM, Ali Pey <[email protected]> wrote: Hello, Let's say I have a sip client that supports both G711 and G729 codecs and I have them both enabled in sip.conf and G729 has higher priority. Can I force the call to choose a different codec based on the dialed number or other conditions? For instance I would want to do G711 if the call was routed to T1 card over Dahdi but G729 if the call was going to another sip client. Thanks, Ali Pey -- _____________________________________________________________________ -- 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
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