| Summary: | ASTERISK-19877: Queued Calls Remain In Queue When Phone In Queue Reboots | ||
| Reporter: | Chris Mylonas (chris.mylonas) | Labels: | |
| Date Opened: | 2012-05-15 13:50:30 | Date Closed: | 2012-05-16 13:26:55 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Applications/app_queue |
| Versions: | 1.8.11.1 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I was doing a load test on some queues and my Internet dropped out (rebooted the router) and the PBX still had the queued calls from when I was on the phone in the queue. I left the calls in the queue. Took up my handset (Snom) and dialed the queue, put the call on hold whilst I waited in the queue, and repeated another several times so the handset had lights flashing - all these calls were waiting in the queue. Rebooted the phone through the settings menu. There are now "15" calls in the queue and when new calls enter the queue, they will never get answered. There must be a way to detect that a phone call is no longer live if the Internet or phone drops out otherwise the queues will just build up. | ||
| Comments: | By: Chris Mylonas (chris.mylonas) 2012-05-15 13:54:59.802-0500 I guess if there was a finite wait time, the queue would die down eventually. Is there a way to detect the call is still there? I will test again tomorrow by not putting calls on hold to see if it only occurs when the person waiting in the queue puts the MoH on hold. It's quite early in the morning here in Syd. Cheers By: Mark Petersen (roadkill) 2012-05-16 06:16:37.856-0500 hi this is not a bug, please use the support forum or IRC for type of question you can use SIP/RTP session timers to detect that the call is gone is sip.conf rtptimeout rtpholdtimeout session-expires By: Chris Mylonas (chris.mylonas) 2012-05-16 09:23:24.653-0500 Hi Mark, Thanks for you response, I should have got back earlier with the same message. A kind gentleman on the freeswitch mailing list suggested I look at sip session timers :) By default I think asterisk uses "accept" when I was after "originate" for the sic "session-timers" setting - I haven't gotten to the bottom of it, but I think it could be because I am using SIPp for my static agents and my scenario.xml file is rather simple. I spent an hour or two referring to the RFC for session timers which was cool. Happy bug fixing, Chris | ||