[asterisk-users] UDP miss a hangup on SIP

[Johan Wilfer]

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