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? Thanks, Jerry
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