Hi, I have a case where SIP channels will not be destroyed, resulting in further calls to ChanIsAvail() to fail. The process (I believe) to replicate this is the following:
- Make a call to another SIP phone that is an "intercom" call (Auto-Answer) - For whatever reason, the phone happens to go UNREACHABLE during this call - Phone comes back REACHABLE, but channel still exists in "core show channels" As an example, here's 3 stuck calls from today: r...@hades:~# asterisk -rx "core show channels" Channel Location State Application(Data) SIP/6296-a2298 (None) Up AppDial((Outgoing Line)) SIP/6315-a0906 *806...@ext-in Up Dial(SIP/6296|5|A(beep)) SIP/6333-a131e (None) Up AppDial((Outgoing Line)) SIP/6294-a24fc *806...@ext-in Up Dial(SIP/6333|5|A(beep)) SIP/6297-a1cb7 (None) Up AppDial((Outgoing Line)) SIP/6315-adc5d *806...@ext-in Up Dial(SIP/6297|5|A(beep)) .... I don't know if this has been fixed in a later 1.4.x version, though after reading some of the DTMF relaying problems with 1.4.27 and beyond, I don't think I would want to upgrade... Thanks. -- James -- _____________________________________________________________________ -- 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
