Anthony,<br><br>Thanks for answering...and sorry for the delay. I've already checked all of the ptime settings I can, and all phones plus freeswitch are set to use 20ms packetization. I've even set "scrooge" in the codec negotiation, but I keep running into this issue. I've updated my post with "sofia global siptrace on".<br>
<br>I am assuming that the ptime issue happens around line 2462 (<a href="http://pastebin.freeswitch.org/16935">http://pastebin.freeswitch.org/16935</a>)<br><br><ol><li class="li2"><div class="de2"><span class="re0"><div style="color: rgb(255, 255, 85); background-color: black;">
<span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-01</span> <span class="nu0">09</span>:<span class="nu0">12</span>:<span class="nu0">37.332892</span> <span class="br0">[</span>DEBUG<span class="br0">]</span> sofia_glue.c:<span class="nu0">4711</span> Audio Codec Compare <span class="br0">[</span>PCMU:<span class="nu0">0</span>:<span class="nu0">8000</span>:<span class="nu0">20</span>:<span class="nu0">64000</span><span class="br0">]</span>/<span class="br0">[</span>PCMU:<span class="nu0">0</span>:<span class="nu0">8000</span>:<span class="nu0">30</span>:<span class="nu0">64000</span><span class="br0">]</span></div>
</span></div></li><li class="li1"><div class="de1"><span class="re0"><div style="color: rgb(255, 255, 85); background-color: black;"><span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-01</span> <span class="nu0">09</span>:<span class="nu0">12</span>:<span class="nu0">37.332892</span> <span class="br0">[</span>DEBUG<span class="br0">]</span> sofia_glue.c:<span class="nu0">2753</span> Already using PCMU</div>
</span></div></li><li class="li2"><div class="de2"><span class="re0"><div style="color: rgb(255, 255, 85); background-color: black;"><span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-01</span> <span class="nu0">09</span>:<span class="nu0">12</span>:<span class="nu0">37.332892</span> <span class="br0">[</span>DEBUG<span class="br0">]</span> sofia_glue.c:<span class="nu0">4819</span> Set <span class="nu0">2833</span> dtmf send payload to <span class="nu0">101</span></div>
</span></div></li><li class="li1"><div class="de1"><span class="re0"><div style="color: rgb(255, 255, 85); background-color: black;"><span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-01</span> <span class="nu0">09</span>:<span class="nu0">12</span>:<span class="nu0">37.332892</span> <span class="br0">[</span>DEBUG<span class="br0">]</span> sofia.c:<span class="nu0">5599</span> Processing updated SDP</div>
</span></div></li><li class="li2"><div class="de2"><span class="re0"><div style="color: rgb(255, 255, 85); background-color: black;"><span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-01</span> <span class="nu0">09</span>:<span class="nu0">12</span>:<span class="nu0">37.332892</span> <span class="br0">[</span>DEBUG<span class="br0">]</span> sofia_glue.c:<span class="nu0">3042</span> Audio params are unchanged for sofia/internal/sip:<span class="nu0">1003</span>@<span class="nu0">10.0</span><span class="nu0">.1</span><span class="nu0">.205</span>:<span class="nu0">5060</span>.</div>
</span></div></li><li class="li1"><div class="de1"><span class="re0"><div style="color: rgb(255, 255, 85); background-color: black;"><span class="nu0">2011</span><span class="nu0">-08</span><span class="nu0">-01</span> <span class="nu0">09</span>:<span class="nu0">12</span>:<span class="nu0">37.332892</span> <span class="br0">[</span>DEBUG<span class="br0">]</span> sofia_glue.c:<span class="nu0">3052</span> sofia/internal/sip:<span class="nu0">1003</span>@<span class="nu0">10.0</span><span class="nu0">.1</span><span class="nu0">.205</span>:<span class="nu0">5060</span> Setting audio receive payload in Re-INVITE to <span class="nu0">0</span></div>
</span></div></li></ol>Could this be an issue with the Aastra'a firmware? Or maybe the MOH is being processed at 30ms instead of 20ms, and the negotiation is not updated somehow?<br><br>I don't mean to sound ignorant, but I'm really at a loss here...and thanks again for any help!<br>
<br>Cheers,<br>Chris<br><br><div class="gmail_quote">On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">probably ptime related thing.<br>
you should have included the sip trace "sofia global siptrace on"<br>
<div><div></div><div class="h5"><br>
<br>
On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <<a href="mailto:cmcureau@gmail.com">cmcureau@gmail.com</a>> wrote:<br>
> I'm having some issues with extremely choppy audio after a call has been<br>
> sent to another extension via an automated transfer. The audio is great<br>
> when the call is answered. Shortly after, the transfer button is pressed<br>
> and the incoming call hears music on hold. The music on hold is sent to the<br>
> caller sounds fine as does the conversation between extensions. When the<br>
> transfer is completed, the caller hears what sounds like someone speaking<br>
> through a fan (though slower) but incoming audio sounds fine.<br>
><br>
> Since it's such a large log, I posted it to the FreeSWITCH pastebin:<br>
> <a href="http://pastebin.freeswitch.org/16911" target="_blank">http://pastebin.freeswitch.org/16911</a><br>
><br>
> I'm thinking that it has something to do with the transition from MOH to the<br>
> internal extension, but I can't figure out what is happening.<br>
><br>
> Any ideas?<br>
><br>
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