Re: [asterisk-users] FXO ports locking up
On Fri, Jul 08, 2011 at 10:58:06AM -0400, Shawn L wrote: > I have a situation where I have an Asterisk box which receives 8 > analog lines from a > Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a > call coming in > on port 1 of the digium FXO board is delivered to SIP phone 1, an > outgoing call on SIP > phone 2 goes out FXO line 2, etc. > > This works fine normally, but every once in a while (no set time, or > pattern that I can > see -- It may be caused by the wifi sip phone going out of range of an > access point and > not coming back into range fast enough) the FXO port does not hangup > after the call is > terminated and just sits in an in-use state. Since it's a 1-to-1 > mapping, the SIP phone > associated with the in-use line now produces a fast busy when you > attempt to make a > call because it cannot get an outbound line. > > 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. look for 'busydetect' in chan_dahdi.conf . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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
Re: [asterisk-users] FXO ports locking up
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/cordless8-04 725@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/cordless8-049c Indirect Bridge: SIP/cordless8-049c -- 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/cordless8-049c DIALEDPEERNUMBER=8/725 TRANSFERCAPABILITY=SPEECH On Fri, Jul 8, 2011 at 7:25 PM, Alec Davis 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
Re: [asterisk-users] FXO ports locking up
> 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
