AFAIK port zero is a method of putting the call on hold in a reinvite, and shouldn&#39;t be in the initial invite.<div><br></div><div>Without a port you can only receive audio not send it (as you have nowhere to send to).</div>
<div><br></div><div>Your SDP shows the phone is hiding its model - what is it? There&#39;s a 2011 mailing list  thread suggesting it may be an issue with a bad firmware on a type of phone.<span></span></div><div><br></div>
<div><br></div><div><br><br>On Monday, August 19, 2013, Vallimamod ABDULLAH  wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">
<div>Hi,</div><div><br></div><div>It looks like the remote SDP does not contain correct rtp port information. You have:</div><div><br></div><div><div>m=audio 0 RTP/AVP 0 18 101</div><div><br></div><div>The first &#39;0&#39; should be the rtp port normally. I am not sure but it does not looks correct and may be the cause of the incompatible destination error.</div>
<div>Hope this helps.</div><div><br></div><div>-- </div><div>Best Regards,</div><div>Vallimamod</div><div>.</div><div><br></div><div><br></div></div>
<br><div><div>On Aug 19, 2013, at 2:59 PM, Royce Mitchell III &lt;<a>royce3@gmail.com</a>&gt; wrote:</div><br><blockquote type="cite"><div dir="ltr"><div style="font-family:courier new,monospace">Here is one example, thanks</div>
</div><div><br clear="all"><div><div><span style="font-family:arial;font-size:small"><br>
</span></div><div><span style="font-family:arial;font-size:small"><br></span></div><span style="font-family:arial;font-size:small">Royce Mitchell, </span>IT Consultant<div style="font-family:arial;font-size:small">ITAS Solutions</div>

<div style="font-family:arial;font-size:small"><a>royce3@itas-solutions.com</a></div></div>
<br><br><div>On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <span dir="ltr">&lt;<a>steveayre@gmail.com</a>&gt;</span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div dir="ltr">Can you paste a debug log of the entire lifetime of the call?</div><div><div><div><br><br><div>On 16 August 2013 21:57, Royce Mitchell III <span dir="ltr">&lt;<a>royce3@gmail.com</a>&gt;</span> wrote:<br>


<blockquote style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div style="font-family:courier new,monospace">The only thing I&#39;m doing in FS regarding codecs is I force PCMU in certain conditions ( ran into a transcoding bug in FreeSwitch between G722 HD and G711 with the Polycom phones )</div>




</div><div><div><br clear="all"><div><div><span style="font-family:arial;font-size:small"><br></span></div><div><span style="font-family:arial;font-size:small"><br></span></div><span style="font-family:arial;font-size:small">Royce Mitchell, </span>IT Consultant<div style="font-family:arial;font-size:small">




ITAS Solutions</div><div style="font-family:arial;font-size:small"><a>royce3@itas-solutions.com</a></div></div>
<br><br></div><div><div>On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <span dir="ltr">&lt;<a>lconroy@insensate.co.uk</a>&gt;</span> wrote:<br>


<blockquote style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
Hi there,<br>
 Are you sure about fS allowing PCMU?<br>
According to the remote SDP, your incall is proposing PCMU as its top choice; g729 is 2nd best (quite right too :).<br>
Forcing the adtran to offer only PCMU should not make ay difference to that -- it&#39;ll still propose PCMU so no change.<br>
Either your setup is somehow blocking PCMU on the b-leg (but I&#39;d expect to see that on the log), OR is trying to transcode because the b-leg requires some (non-PCMU) codec and can&#39;t (again, I&#39;d expect that to be logged), OR fS is not accepting PCMU.<br>





Assuming that PCMU is in the fS vars codec lists, does your dialplan do anything funky with the codec list for an incall?<br>
all the best,<br>
  Lawrence<br>
<div><br>
On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:<br>
&gt; My FreeSWITCH is configured to prefer PCMU, and the devices it is talking<br>
&gt; to are Adtran 908e&#39;s. The Adtrans are configured for the default codec<br>
&gt; group which is supposed to be PCMU, but I can reconfigure them to<br>
&gt; explicitly allow only PCMU. I will try that and see if it makes a<br>
&gt; difference.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Royce Mitchell, IT Consultant<br>
&gt; ITAS Solutions<br>
&gt; <a>royce3@itas-solutions.com</a><br>
&gt;<br>
&gt;<br>
&gt; On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre &lt;<a>steveayre@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; INCOMPATIBLE_DESTINATION means a codec problem.<br>
&gt;&gt;<br>
&gt;&gt; The remote SDP sends they&#39;re offerring PCMU and G729.<br>
&gt;&gt;<br>
&gt;&gt; What codecs are you allowing, what codecs are you bridging with, and since<br>
&gt;&gt; G729 is on the list are you perhaps trying to transcode without using<br>
&gt;&gt; mod_com_g729+licenses?<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On 16 August 2013 16:07, Royce Mitchell III &lt;<a></a></div></blockquote></div></div></div></blockquote></div></div></div></div></blockquote></div></div></blockquote></div></div></blockquote></div>