Doesn't seem to help. I did it early yesterday morning and have another 'stuck' call this morning
Does anyone have any other ideas on what I can do to correct this? thanks Shawn CLI> core show channels Channel Location State Application(Data) DAHDI/8-1 (None) Up AppDial((Outgoing Line)) SIP/cordless8-000004 725@out-phone8:1 Up Dial(DAHDI/8/725) 2 active channels 1 active call CLI> core show channel DAHDI/8-1 -- General --> Name: DAHDI/8-1 Type: DAHDI UniqueID: 1310421996.2359 Caller ID: 725 Caller ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 23 Frames in: 2489590 Frames out: 72966 Time to Hangup: 0 Elapsed Time: 13h49m51s Direct Bridge: SIP/cordless8-0000049c Indirect Bridge: SIP/cordless8-0000049c -- PBX -- Context: in-phone8 Extension: Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Blocking in: ast_waitfor_nandfds Variables: BRIDGEPVTCALLID=2e52745c-7bdfef53@192.168.0.134 BRIDGEPEER=SIP/cordless8-0000049c DIALEDPEERNUMBER=8/725 TRANSFERCAPABILITY=SPEECH On Fri, Jul 8, 2011 at 7:25 PM, Alec Davis <siva...@paradise.net.nz> wrote: >> Is there a way to detect that there is no longer really an >> active call happening and force a hangup or reset the >> channel? It'd be great if this could happen automatically. >> Or as a temporary fix , is there a way to setup and extension >> that the SIP phone could dial which would clear any active >> calls associated with it? Right now if this happens, I need >> to login to the Asterisk CLI and issue a hangup command. If >> I don't, the channel appears to be in-use forever. > > This may be the answer > > sip.conf: > > ;--------------------------- RTP timers > ---------------------------------------------------- > ; These timers are currently used for both audio and video streams. The RTP > timeouts > ; are only applied to the audio channel. > ; The settings are settable in the global section as well as per device > ; > rtptimeout=60 ; Terminate call if 60 seconds of no RTP or > RTCP activity > ; on the audio channel > ; when we're not on hold. This is to be able > to hangup > ; a call in the case of a phone disappearing > from the net, > ; like a powerloss or grandma tripping over > a cable. > > Alec Davis > > > -- > _____________________________________________________________________ > -- 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