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
[EMAIL PROTECTED] [EMAIL PROTECTED] 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
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