Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1.
It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the logging output and found the following [Dec 15 11:39:37] VERBOSE[10419] logger.c: -- Packet2Packet bridging SIP/tel01-b6b09b18 and SIP/spa941_0027-09047cf8 [Dec 15 11:40:45] DEBUG[31007] app_queue.c: Device 'SIP/spa941_0027' changed to state '3' (Busy) [Dec 15 11:41:40] DEBUG[10481] app_queue.c: SIP/spa941_0027 in use, can't receive call [Dec 15 11:42:43] DEBUG[31007] app_queue.c: Device 'SIP/spa941_0027' changed to state '3' (Busy) [Dec 15 11:45:18] DEBUG[31008] chan_sip.c: Destroying user object from memory: spa941_0027 [Dec 15 11:45:41] DEBUG[10619] app_queue.c: SIP/spa941_0027 in use, can't receive call [Dec 15 11:45:52] DEBUG[10626] app_queue.c: SIP/spa941_0027 in use, can't receive call [Dec 15 11:46:39] DEBUG[31008] chan_sip.c: Allocating new SIP dialog for [email protected] - REGISTER (No RTP) [Dec 15 11:46:39] DEBUG[31007] app_queue.c: Device 'SIP/spa941_0027' changed to state '1' (Not in use) As you can see - the agent with spa941_0027 does have an active call starting at 11:39:37 - it does get marked as busy (because of call limit) - thats correct. At 11:45:18 there was a sip reload - the user object gets destroyed - but the peer object not - so the busy level is still correct. Than at 11:46:39 the sip phone does reregister at the system - and the system does change the peer to be marked as not in use - from this point things are going wrong.... So i think the way to reproduce is - "active call" -> "sip reload", "reregister", "not in use state" I have to verify this to be reproduceable - but wanted to ask here firstly if someone does already know this behaviour... I have seen bug http://bugs.digium.com/view.php?id=13525 - i think it is releated to it Here are the relevant sip settings Realtime SIP Settings: ---------------------- Realtime Peers: Yes Realtime Users: Yes Cache Friends: Yes Update: Yes Ignore Reg. Expire: No Save sys. name: Yes Auto Clear: 120 Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 360 secs regards, Wolfgang _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
