2012-08-16 02:13, Jerry Geis skrev:
Is it possible to miss a UDP SIP packet to hangup a call?
Using 1.4.43 I had a call from on asterisk box (server) to a
low end client (chan_alsa) not hangup.
Could this be due to missed UDP SIP packet to hangup?
Is there anyway for a client asterisk (chan_alsa again) to
monitor the connection and if the channel is not there to
hangup?
In sip.conf you could use rtp-timers to hangup a call if the
media-stream stops to flow.
Look at these options in sip.conf:
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=0
--
Johan Wilfer
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