Yes, G.729a and G.729b are incorrect and your device is at fault... only other phones of the same type would probably recognise it without issue.<br><br>PCMA should be used in this case though.<br><br>RTP payload numbers are spread through multiple RFCs. Anything in the 96-127 range is dynamic and the codec is determined from the matching rtpmap line, any of the static numbers don't need a rtpmap line to work. IANA oversees assignment of the static numbers and they have the full list: <a href="http://www.iana.org/assignments/rtp-parameters">http://www.iana.org/assignments/rtp-parameters</a><br>
<br>As you can see 0=PCMU, 8=PCMA, and G.729 should use 18. Support of annex B is specified in the fmtp parameter, not the codec name - e.g. "a=fmtp:18 annexb=no". Annex A never needs to be specified as it can be read normally by plain G.729, so it's just up to the implementation on whether it wants to save quality or cpu when encoding.<br>
<br>Do you have any other applications running which would also be using the RTP port range? A call will fail if it tries to use a port that's already in use, perhaps with that message. FS should avoid using ports it's already using, but can't know about any other programs on the system.<br>
<br>-Steve<br><br><br><div class="gmail_quote">On 27 August 2010 08:03, Jonas Gauffin <span dir="ltr"><<a href="mailto:jonas.gauffin@gmail.com">jonas.gauffin@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div>It doesn't happen every time and it's on a production system with a bit of volume. therefore a bit hard to get SIP traces. I'll try if Anthony really needs them.</div>
<div> </div>
<div>FS do say this: </div>
<div><span>2010</span><span>-08</span><span>-27</span> <span>07</span>:<span>14</span>:<span>10.758750</span> <span>[</span>DEBUG<span>]</span> switch_ivr_originate.c:<span>3111</span> sofia/external/<span>0700123456</span>@<span>212.151</span>.Y.Y:<span>5060</span> Media Establishment Failed</div>
<div> </div>
<div>In which RFC are codec names defined? rfc3551 defines "G729" but no "G.729A" or "G.729B". But as you say, shouldn't FS use PCMA in any case?</div><div><div></div><div class="h5">
<div><br><br> </div>
<div class="gmail_quote">On Fri, Aug 27, 2010 at 8:43 AM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;" class="gmail_quote">Ordinarily it'd mean there was a problem with the codecs (e.g. needing to use an unsupported codec or transcode a codec that only works as a passthrough one).<br>
<br>Looks like it should have gone through with PCMA (8) though. Can you repeat the call with sip trace on? Perhaps the incompatible destination comes from an endpoint.<br><br>'sofia profile <profilename> siptrace on' from the CLI, replace on with off to turn it off again.<br>
<br>-Steve<br><br><br><br>
<div class="gmail_quote">
<div>
<div></div>
<div>On 27 August 2010 07:26, Jonas Gauffin <span dir="ltr"><<a href="mailto:jonas.gauffin@gmail.com" target="_blank">jonas.gauffin@gmail.com</a>></span> wrote:<br></div></div>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;" class="gmail_quote">
<div>
<div></div>
<div>
<div>Hello,</div>
<div> </div>
<div>Why do this call result in INCOMPATIBLE_DESTINATION?</div>
<div> </div>
<div><a href="http://pastebin.freeswitch.org/13736" target="_blank">http://pastebin.freeswitch.org/13736</a></div>
<div> </div>
<div>Regards,</div>
<div> Jonas</div><br></div></div>
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