Ok, I have posted a siptrace of the call failing. If you need more information, just let me know and I'll get what I can. I cleaned up domain names and CallerID names, but everything else should be untouched. The link is<div>
<a href="http://pastebin.freeswitch.org/19886">http://pastebin.freeswitch.org/19886</a></div><div><br></div><div>Thanks again.</div><div><br></div><div>Carlos<br><br><div class="gmail_quote">On Tue, Sep 11, 2012 at 7:53 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Can you collect console logs w/ SIP trace? Drop them on <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a> and link back here. I think we have a few Polycom gurus who may have some experience here, but they probably will need more info before they can help.<br>
<br>-MC<br><br><div class="gmail_quote"><div><div class="h5">On Tue, Sep 11, 2012 at 11:43 AM, Carlos Flor <span dir="ltr"><<a href="mailto:jackal@cybershroud.net" target="_blank">jackal@cybershroud.net</a>></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 class="h5">
<p>
</p><p>So here is the situation: I have a polycom phone with extension 101@pbxA and 101@pbxB. Someone else has a polycom phone with extension 102@pbxA and 102@pbxB. If I try to call from 101@B to 102@B, the phone rings, but as soon as 102 answers, the call hangs up immediately. If I change 102@B to 103@B, so that the extensions are now 102@A and 103@B on the second phone, then phone calls from the second phone to the first work, but not the other way around. If I repeat the change on the first phone, so that it now has 101@A and 104@B, then calls work correctly in both directions.</p>
<p>So, it seems as though if you have more than one registration but use the same extension on each, the polycom has issues with the RTP piece of the call. The SIP piece seems to work ok because the call actually makes it to the other phone and rings, but as soon as you pickup (when RTP should start) the call ends.</p>
<p>Has anyone run into anything close to this before? I am sure my description is confusing and it's much easier to explain on a whiteboard, but hopefully it makes sense.</p><p>Just to clarify, the two PBXs are not related to each other. I'm not trying to call from pbxA to pbxB.</p>
<span><font color="#888888">
<p><br></p><p>Carlos</p><p></p>
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