Hi, Am Dienstag, den 16.12.2014, 16:32 +0100 schrieb Karsten Wemheuer: > Hi, > > I got a weird behaviour in asterisk (original found in 1.8 but it is > still the same in 11.15.0). I have three phones communicating via > OpenSIPs with asterisk. Phone A dials 100 and asterisk calls > SIP/phone-b. Phone B accepts the call. The User on Phone B places the > call on hold, dials 200 and, while hearing the dial tone of ringing > Phone C, places the handset on hook. Phone B sends a REFER, so that > Phone A is connected with the ringing Phone C. Asterisk sends an UPDATE > to Phone-C to update the connected line information. Now the user on > Phone B realized that User B is not available. He presses the blinking > LED of BLF to get the Call back. A and B are now connected again. But > Phone C is still ringing. Asterisk sends the CANCEL to terminate the > call to phone C not to the proxy (where the INVITE comes from), but > directly to the phone. The phone ignores this CANCEL, as it does not > belong to a call and so the phone keeps on ringing. > > If I modify the configuration, so that there is no UPDATE for the > connected line information, the CANCEL is send via proxy to the phone > and all is well.
I possibly find the source of the failure and a patch, working without failure in my test scenario. I filed a bug with patch attached, see https://issues.asterisk.org/jira/browse/ASTERISK-24628 Have a nice day, Karsten -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
