Your call is up on VoiceMail you should check dialstatus before sending user to VoiceMail.
On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain <[email protected]> wrote: > This just started after upgrading to 11.11.0. After a call is > completed (both ends hang up) the call still shows as active. > > # asterisk -x "core show channels" > Channel Location State Application(Data) > SIP/thinktel-0000000 (None) Up AppDial((Outgoing > Line)) SIP/4164251212-00000 4165555555@LocalSets Up > Dial(SIP/thinktel/4165559999) 2 active channels > 1 active call > 1 call processed > > The 1212 number is mine and is hung up. I even rebooted my ATA to make > sure that it wasn't holding the line. My dialplan is extremely > simple. In fact, I even simplified it from what it was for this > testing. Here it is. > > exten => 4164251212,1,Verbose(0, ${CALLERID(all)} Calling ${EXTEN}) > same => n,Dial(SIP/4164251212,30) > same => n,VoiceMail(4164251212@LocalSets,u) > same => n,Hangup() > > I can post any other log or config excerpts if someone thinks that they > are relevant but all of this was working under 11.10.2. > > Thanks. > > > -- > D'Arcy J.M. Cain > System Administrator, Vex.Net > http://www.Vex.Net/ IM:[email protected] > VoIP: sip:[email protected] > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
