Summary: | ASTERISK-06512: SIP Callers to Queue Get Dumped | ||
Reporter: | damin (damin) | Labels: | |
Date Opened: | 2006-03-09 13:01:26.000-0600 | Date Closed: | 2011-06-07 14:03:18 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
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; [support] music=default strategy=roundrobin timeout=2 retry=2 wrapuptime=2 maxlen=0 announce-frequency=10 announce-holdtime=yes announce-round-seconds=10 context=operator 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; [outbound] ; Hangup if we Timeout exten => t,1,Hangup include => queues ; Our Call Queues [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 207.166.192.127:11837: CANCEL sip:2@207.166.192.186;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.51;branch=z9hG4bK8908fec097B3E025 From: "Gregory Boehnlein" <sip:100@207.166.192.186>;tag=BC4F44CF-7A545902 To: <sip:2@207.166.192.186;user=phone>;tag=as17a74d47 CSeq: 2 CANCEL Call-ID: b416d413-d4668549-1195f1d4@192.168.2.51 Contact: <sip:100@192.168.2.51> Proxy-Authorization: Digest username="100", realm="asterisk", nonce="0b47b602", uri="sip:2@207.166.192.186;user=phone", response="6448a8a407e1f8e07550a1ade49c2d95", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 |