<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
code
{mso-style-priority:99;
font-family:"Courier New";}
span.hoenzb
{mso-style-name:hoenzb;}
span.EmailStyle20
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-family:"Calibri","sans-serif";}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP_CODEC is only useable on a SIP channel. You can specify DAHDI codecs in users.conf.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Ali Pey<br><b>Sent:</b> Friday, November 02, 2012 12:28 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Different codec for different type of calls<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Qasim,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thank you for your response. I tried it but still doesn't work. This is what I have:<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>exten => _XXX.,1,NoOP(Set G711 codec)<o:p></o:p></p></div><div><p class=MsoNormal>exten => _XXX.,n,Set(SIP_CODEC=ulaw)<o:p></o:p></p></div><div><p class=MsoNormal>exten => _XXX.,n,Set(SIP_CODEC_OUTBOUND=ulaw)<o:p></o:p></p></div><div><p class=MsoNormal>exten => _XXX.,n,Dial(DAHDI/g1/$EXTEN)<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Then I get this error:<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>WARNING[12156]: channel.c:5796 ast_request: No translator path exists for channel type DAHDI (native (ulaw|alaw|slin)) to (h264|silk8)<o:p></o:p></p></div><div><p class=MsoNormal>WARNING[12156]: app_dial.c:2277 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available)<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I tried both g711 and ulaw with no luck. I read the SIP_CODEC value and it is set properly. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Any suggestions/ideas?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thanks,<o:p></o:p></p></div><div><p class=MsoNormal>Ali Pey<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Thu, Nov 1, 2012 at 12:02 PM, <a href="mailto:qasimakhan@gmail.com">qasimakhan@gmail.com</a> <<a href="mailto:qasimakhan@gmail.com" target="_blank">qasimakhan@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal><code><span style='font-size:10.0pt'>exten => _X.,1,NoOP(G711 CoDec)</span></code><span style='font-size:10.0pt;font-family:"Courier New"'><br><code>exten => _X.,n,Set(SIP_CODEC=g711)</code><br><code>exten => _X.,n,Dial(...)</code><br><br></span><strong>${SIP_CODEC}</strong>: 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 <a href="https://issues.asterisk.org/view.php?id=13243" target="_blank">SIP_CODEC_OUTBOUND</a> for the remote (=second) call leg.<br><br>Hope this helps,<br><br>Regards,<br>Qasim<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><div><div><p class=MsoNormal>On Thu, Nov 1, 2012 at 8:49 PM, Ali Pey <<a href="mailto:alipey@gmail.com" target="_blank">alipey@gmail.com</a>> wrote:<o:p></o:p></p></div></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><p class=MsoNormal>Hello,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>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.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Can I force the call to choose a different codec based on the dialed number or other conditions?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>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.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thanks,<o:p></o:p></p></div><div><p class=MsoNormal>Ali Pey<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal><span class=hoenzb><span style='color:#888888'>--</span></span><span style='color:#888888'><br><span class=hoenzb>_____________________________________________________________________</span><br><span class=hoenzb>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --</span><br><span class=hoenzb>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span class=hoenzb> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a></span><br><br><span class=hoenzb>asterisk-users mailing list</span><br><span class=hoenzb>To UNSUBSCRIBE or update options visit:</span><br><span class=hoenzb> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></span><o:p></o:p></p></blockquote></div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>