|Summary:||ASTERISK-04817: Asterisk doesn't detect a device disconnect while in a call|
|Date Opened:||2005-08-11 10:01:55||Date Closed:||2011-06-07 14:10:20|
|Description:||Suppose you have a call between two local SIP devices, SIP1 -> Asterisk -> SIP2. SIP2 places the call on hold and music on hold starts playing on SIP1. SIP2 disconnects (powers off, the netwok cable is disconnected, the device reboots, etc). The call will stay on hold indefinitely (well, at least to the limit of my patience). SIP1 has no way of knowing that the other party is no longer there. I can provide traces, but I believe that the problem is pretty easily reproduced. I tested with today's HEAD (8/11/05)|
|Comments:||By: Olle Johansson (oej) 2005-08-11 10:10:01|
Test the rtp timeout settings in sip.conf to fix this.
By: mihai (mihai) 2005-08-11 10:33:44
Thanks, that pretty much solved the problem. However, there's still an issue with devices that don't send anything when on hold. Is there any plan to address this at SIP level (one way of doing this would be to send OPTIONS requests periodically)
By: Olle Johansson (oej) 2005-08-11 13:10:47
Well, if you look around there's a patch for SIP timers in the bug tracker. It does not solve the timeouts, but is a first step towards an implementation of the SIP timer extension. We first need the basic timers to work properly.
Will close this as resolved.
By: Olle Johansson (oej) 2005-08-11 13:11:16
Configuration issue, not a bug.