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
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