[asterisk-users] FXO ports locking up

[Shawn L]

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 at 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 at 192.168.0.134
BRIDGEPEER=SIP/cordless8-0000049c
DIALEDPEERNUMBER=8/725
TRANSFERCAPABILITY=SPEECH

On Fri, Jul 8, 2011 at 7:25 PM, Alec Davis <sivad.a at 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
>
>
> --
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