[asterisk-users] Channels hanging when SIP phone gets reset during call

[Olle E Johansson]

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