asterisk users - Feb 2007 - Channels hanging when SIP phone gets reset during call

Hi All.

This is on Asterisk 1.2.13

I place a call between 2 SIP phones (with canreinvite=yes, qualify=yes).

I reset the phones (so they don't have time to say BYE).

Asterisk seems to think that the call is still ongoing. This persists
until I do a 'restart now'.

asterisk1*CLI> show channels
Channel              Location             State   Application(Data)
SIP/5301-089fc890    (None)               Up      Bridged
Call(SIP/5303-089f1558
SIP/5303-089f1558    s@macro-dial:10      Up      Dial(SIP/5301||)
2 active channels
1 active call

asterisk1*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold
Last Message
192.168.5.203    line0       2eeb3516264  00103/00000  ulaw  No
Tx: ACK
192.168.5.203    5303        28948-0xca0  00102/00001  ulaw  No
Tx: ACK
2 active SIP channels


asterisk1*CLI> show hints
asterisk1*CLI>
    -= Registered Asterisk Dial Plan Hints =-
   5303                : SIP/5303              State:InUse
Watchers  1
   5301                : SIP/5301              State:InUse
Watchers  0
----------------
- 2 hints registered

I was wondering whether there is anything that I can do about this on
either the Asterisk server or the phone?

______________________________
Steve Langstaff

21 feb 2007 kl. 12.54 skrev Steve Langstaff:
> Hi All.
>
> This is on Asterisk 1.2.13
>
> I place a call between 2 SIP phones (with canreinvite=yes,  
> qualify=yes).
>
> I reset the phones (so they don't have time to say BYE).
>
> Asterisk seems to think that the call is still ongoing. This persists
> until I do a 'restart now'.
Check the RTP timers in sip.conf. They will hangup the call if there's
no audio.

/O