I found out what the problem was.<br><div class="gmail_quote"><br>It appears to be a bug in the Polycom 430 firmware.<br><br>I have 2 lines on the phone and both of them use the same auth id but with different servers.<br>
<br>It seems that if you make an outgoing call from the phone on line 2 and then called party hangs up. Asterisk says BYE and the Polycom looks at line 1 (because it has the same auth id as line 2) and says I don't have an active call on line 1 when the active call is on line 2.<br>
<br>Kinda annoying, but easy enough to work around.<br><br>I am in the middle of migrating systems so I can just change all my usernames on line 2 to be prefixed with 1 or something like that.<div><div></div><div class="Wj3C7c">
<br><br><div class="gmail_quote">
On Mon, Nov 3, 2008 at 11:16 AM, Joel Pearson <span dir="ltr"><<a href="mailto:joel.pearson%2Basterisk@gmail.com" target="_blank">joel.pearson+asterisk@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;">
Hi,<br><br>I have a really strange problem with a Polycom 430 phone and Asterisk<br><a href="http://1.4.20." target="_blank">1.4.20.</a><br><br>Currently If I dial the Polycom from my mobile phone answer the call on the<br>
Polycom and then hangup the mobile the call ends fine on the Polycom.<br>
But if I call from the Polycom to my mobile and then I hang up the mobile<br>the Polycom thinks the call is still active.<br><br>However doing a show sip channels shows the the call has ended.<br><br>Further to that doing a tcpdump shows that Asterisk sends a SIP BYE to the<br>
phone but the phone responds with:<br>Status 481 Call Leg/Transaction does not exist.<br><br>The Polycom is currently associated with 2 sip servers (using 2 lines on the<br>phone) because I am currently in the progress of migrating from one server<br>
to another.<br><br>So the asterisk server is having issues with is on Line 2 and it works<br>perfectly well on Line 1 with a completely different Asterisk server running<br><a href="http://1.4.16.2" target="_blank">1.4.16.2</a>.<br>
<br>I haven't tried switching the lines around to see if its just a problem with<br>
it being on Line 2.<br><br>The Polycom is running the latest Bootrom and Sip version.<br><br>Does anyone have any idea what could be causing this?<br><br>Cheers,<br><font color="#888888"><br>-Joel<br>
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