Thanks, yes, here you go:<div><br></div><div><div>I'm making calls from the console to an X-Lite registered on extension 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec options are not supported, and even though on the third call the absolute_codec_string variable should be 'G729,PCMU', it is not offering both codecs.</div>
<div><br></div><div><br></div><div>On <a href="http://pastebin.freeswitch.org/16167">http://pastebin.freeswitch.org/16167</a> the SIP trace for:</div><div><br></div><div>originate {ignore_early_media,absolute_codec_string=G729}user/1001 &bridge(user/1000)</div>
<div><br></div><div>This is the SDP:</div><div><br></div><div> v=0</div><div> o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1</div><div> s=FreeSWITCH</div><div> c=IN IP4 127.0.0.1</div><div> t=0 0</div><div>
m=audio 24036 RTP/AVP 18 101 13</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=ptime:20</div><div><br></div><div><br></div><div>On <a href="http://pastebin.freeswitch.org/16169">http://pastebin.freeswitch.org/16169</a> the SIP trace for:</div>
<div><br></div><div>originate {ignore_early_media,absolute_codec_string=PCMU}user/1001 &bridge(user/1000)</div><div><br></div><div>This is the SDP:</div><div><br></div><div> v=0</div><div> o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1</div>
<div> s=FreeSWITCH</div><div> c=IN IP4 127.0.0.1</div><div> t=0 0</div><div> m=audio 24896 RTP/AVP 0 101 13</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=ptime:20</div>
<div><br></div><div><br></div><div>On <a href="http://pastebin.freeswitch.org/16170">http://pastebin.freeswitch.org/16170</a> the SIP trace for:</div><div><br></div><div>originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001 &bridge(user/1000)</div>
<div><br></div><div>This is the SDP:</div><div><br></div><div> v=0</div><div> o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1</div><div> s=FreeSWITCH</div><div> c=IN IP4 127.0.0.1</div><div> t=0 0</div><div>
m=audio 25138 RTP/AVP 18 101 13</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=ptime:20</div></div><div><br><br></div><div><br></div><div><br><div class="gmail_quote">On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <span dir="ltr"><<a href="mailto:paul@cupis.co.uk">paul@cupis.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="im">On 25/04/11 18:08, Nicolas Brenner wrote:<br>
> I'm trying to use the<br>
> absolute_codec_string with originate from the console, like so:<br>
><br>
> originate<br>
> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444<br>
> &bridge(user/1001)<br>
><br>
> Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same as<br>
> if I don't quote the string, or if I just specify one codec:<br>
<br>
</div>Can you provide (on <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a>) a complete log of a call,<br>
please?<br>
<div><div></div><div class="h5"><br>
Regards,<br>
<br>
<br>
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