I have asked this question elsewhere (other forums), I hope that’s not a violation of anything nor offensive to those who’ve offered help, I am just trying to cast a wide net.
In this scenario I am using a sip.js browser-based softphone which works perfectly into pjsip via web sockets.
If I have an ongoing call which is terminated for reason of network connection failure, asterisk picks it up almost straight away with something like:
[Aug 28 21:20:26] WARNING: res_http_websocket.c:523 ws_safe_read: Web socket closed abruptly -- Removed contact 'sip:email@example.com.XX.XXX:41453;transport=ws' from AOR 'adrian' due to shutdown == Contact adrian/sip:firstname.lastname@example.org.XX.XXX:41453;transport=ws has been deleted == Endpoint adrian is now Unreachable == WebSocket connection from '86.18.XX.XXX:41453' closed
However, the call itself is not terminated, it just ticks along happily in the background until I manually issue a hangup request.
How can I make asterisk disconnect the call when one of the endpoints participating in it drops offline?
I have tried lowering timers_min_se and timers_sess_expires as low as they’ll go to no avail, surely there is some way to connect these dots - asterisk knows the endooint is gone, why isn’t it throwing away the associated channel?
Thank you all!