Joel Pearson
2008-Nov-03 00:16 UTC
[asterisk-users] Polycom 430 no hangup after SIP BYE, Status 481 instead
Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the call has ended. Further to that doing a tcpdump shows that Asterisk sends a SIP BYE to the phone but the phone responds with: Status 481 Call Leg/Transaction does not exist. The Polycom is currently associated with 2 sip servers (using 2 lines on the phone) because I am currently in the progress of migrating from one server to another. So the asterisk server is having issues with is on Line 2 and it works perfectly well on Line 1 with a completely different Asterisk server running 1.4.16.2. I haven't tried switching the lines around to see if its just a problem with it being on Line 2. The Polycom is running the latest Bootrom and Sip version. Does anyone have any idea what could be causing this? Cheers, -Joel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081103/37fd425f/attachment.htm
Joel Pearson
2008-Nov-07 00:19 UTC
[asterisk-users] Polycom 430 no hangup after SIP BYE, Status 481 instead
I found out what the problem was. It appears to be a bug in the Polycom 430 firmware. I have 2 lines on the phone and both of them use the same auth id but with different servers. 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. Kinda annoying, but easy enough to work around. 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. On Mon, Nov 3, 2008 at 11:16 AM, Joel Pearson < joel.pearson+asterisk at gmail.com <joel.pearson%2Basterisk at gmail.com>> wrote:> Hi, > > I have a really strange problem with a Polycom 430 phone and Asterisk > 1.4.20. > > Currently If I dial the Polycom from my mobile phone answer the call on the > Polycom and then hangup the mobile the call ends fine on the Polycom. > But if I call from the Polycom to my mobile and then I hang up the mobile > the Polycom thinks the call is still active. > > However doing a show sip channels shows the the call has ended. > > Further to that doing a tcpdump shows that Asterisk sends a SIP BYE to the > phone but the phone responds with: > Status 481 Call Leg/Transaction does not exist. > > The Polycom is currently associated with 2 sip servers (using 2 lines on > the > phone) because I am currently in the progress of migrating from one server > to another. > > So the asterisk server is having issues with is on Line 2 and it works > perfectly well on Line 1 with a completely different Asterisk server > running > 1.4.16.2. > > I haven't tried switching the lines around to see if its just a problem > with > it being on Line 2. > > The Polycom is running the latest Bootrom and Sip version. > > Does anyone have any idea what could be causing this? > > Cheers, > > -Joel >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081107/7299792a/attachment.htm
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