We also use Polycom IP650 phones. They are assigned to our customer
service department. Each SIP interface is a member of our customer
service Queue in Asterisk.
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately hanging up will clear
the state.
When this happens there is no SIP channel and the SIP peer appears
normal. I have been unable to isolate a procedure to duplicate the
problem. It happens erratically to all member interfaces throughout the
day. I know that removing the call-limit option from the device's
config will stop the problem. This will also remove the ability for the
SIP channel driver to track the device's state so we can't remove it
permanently.
The phones in question are configured with one line that will except
only one call. The device itself does not think it is in-use because it
will accept another call. Something in the SIP channel driver is not
clearing the state when a call is completed.
There is definitely no correlation between this and Asterisk restarting.
In fact, if a device is 'stuck' on in-use, restarting Asterisk will
clear the state.
I've been working on this for a week now. It only started for us
because I just implemented the call-limit option in the sip.conf in
Asterisk for the devices. See my posts with subject 'Queue and
Interface time out'.
James Andrewartha wrote:
Olle E Johansson wrote:
23 jan 2007 kl. 16.09 skrev Chris Bullock:
I'm running into an issue w/ Buddy status on Polycom IP650 phones using
buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status
on the phones will "stick" in the busy status. I have noticed that I
can call that extension & the status will reset (sometimes). Anyone
else encountered this or anything similar.
I've seen reports on it, but haven't been able to repeat this. I need to
know a way to force this to happen, repeatably. If I can get that, I can
propably trace it and fix it.
It can also happen if you have packet loss in the network, of course.
I've seen it happen when asterisk restarts (or possibly even just reloads
SIP) without the phone being restarted - it's generally accompanied by
-- Incoming call: Got SIP response 500 "Internal Server Error" back from
10.0.0.51
on the console. I think the status gets stuck as "available" most of the
time, but you don't notice it because that's the default.
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