Hi,<div><br></div><div> Is there someone in the list interested in some consulting work to assist me to get to the bottom of this problem?</div><div><br></div><div> Happy to pay for assistance here.</div><div><br></div><div>

Thanks,</div><div><br></div><div>Michael<br><br><div class="gmail_quote">On Thu, Oct 27, 2011 at 11:34 AM, Michael Toop <span dir="ltr">&lt;<a href="mailto:michaelt@callall.co.za">michaelt@callall.co.za</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi,<div><br></div><div> Start &amp; stop RTP ports are commented out in switch.conf so must be the defaults, 16384 &amp; 32768.</div>

<div><br></div><div> I have also tried the NDLB-allow-bad-iananame suggestion &amp; have the same results, see SIP trace here: <a href="http://pastebin.com/vdztr7cf" target="_blank">http://pastebin.com/vdztr7cf</a></div>

<div><br></div><div> What is very strange about this issue it only happens on certain routes i.e. the SDP of the B leg of the call is somehow breaking things.  Really at a loss here.</div><div><br></div><div>Cheers,</div>


<div><br></div><font color="#888888"><div>Michael</div><div><br></div></font><div><br><br><div class="gmail_quote"><div><div></div><div class="h5">On Thu, Oct 27, 2011 at 9:26 AM,  <span dir="ltr">&lt;<a href="mailto:chrisbware@interfree.it" target="_blank">chrisbware@interfree.it</a>&gt;</span> wrote:<br>


</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div class="h5">
<span>Even payload type 96 for G729 sounds strange to me.</span><br><span>How did you set  <span style="white-space:pre-wrap"><em>rtp-start-portand </em><em></em></span></span><br><u></u><em><em>rtp-end-port ?</em></em><span> </span><br>


<span><span style="white-space:pre-wrap"><em></em></span></span><br><u></u><em><em><br></em></em><span> </span><br><span><span style="white-space:pre-wrap"><em></em></span></span><br><u></u><em><em><br></em></em><span> </span><br>


<span>-----Messaggio originale-----</span><br><span>Da: Michael Toop &lt;<a href="mailto:michaelt@callall.co.za" target="_blank">michaelt@callall.co.za</a>&gt;</span><br><span>Inviato il: 26 Ott 2011 - 05:00</span><br><span>A: FreeSWITCH Users Help &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;</span><div>


<br><span><br><br>Hi,</span><br><div> When initiating a call from Freeswitch to a provider, Freeswitch is putting &#39;0&#39;/ zero in the media port.  This looks like a bug to me.:</div><div><span style="font-family:arial, sans-serif;font-size:13px;background-color:#ffffff"><div>


   m=audio 0 RTP/AVP 96 101</div></span></div><div>Media Description, name and address (m): audio 0 RTP/AVP 96 101</div><div> Txt based SIP trace attached (not sure how to scrub pcap files of IPs).</div> <div>Thanks,</div>


<div>Michael</div>

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