|Summary:||ASTERISK-06512: SIP Callers to Queue Get Dumped|
|Date Opened:||2006-03-09 13:01:26.000-0600||Date Closed:||2011-06-07 14:03:18|
|Description:||We have been testing app_queue in house on 1.2 SVN code for a client. We have found that SIP callers to a queue are hungup on consitently after a couple of minutes. IAX2 calls inbound to the queue can stay in the queue until answered. The message that is displayed on the console is:|
"User disconnected from queue support while waiting their turn"
****** ADDITIONAL INFORMATION ******
Here is our queue config. Timings are shortened on it for testing;
member => SIP/100
member => SIP/116
member => SIP/108
member => SIP/112
And our dial-plan logic to call the queue from a SIP extension is pretty simple;
; Hangup if we Timeout
exten => t,1,Hangup
include => queues
; Our Call Queues
exten => 2,1,Queue(support)
|Comments:||By: damin (damin) 2006-03-09 13:34:13.000-0600|
After several tests of this using a stopwatch, we have determined that the SIP channel is hung-up after exactly 60 seconds. Are we bumping up against some time-limit somewhere?
By: damin (damin) 2006-03-09 13:49:23.000-0600
We tracked this down to being a "polycomism". The Polycom will issue a CANCEL after 60 seconds if the call is not answered.
I'm closing this out.
By: damin (damin) 2006-03-09 13:50:26.000-0600
<-- SIP read from 184.108.40.206:11837:
CANCEL sip:firstname.lastname@example.org;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.51;branch=z9hG4bK8908fec097B3E025
From: "Gregory Boehnlein" <sip:email@example.com>;tag=BC4F44CF-7A545902
CSeq: 2 CANCEL
Proxy-Authorization: Digest username="100", realm="asterisk", nonce="0b47b602", uri="sip:firstname.lastname@example.org;user=phone", response="6448a8a407e1f8e07550a1ade49c2d95", algorithm=MD5