How to hangup a call when Endpoint goes unreachable?

[adrianhand]

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:

Code:

[Aug 28 21:20:26] WARNING[22370]: res_http_websocket.c:523 ws_safe_read: Web socket closed abruptly
-- Removed contact 'sip:pehgks63@86.18.XX.XXX:41453;transport=ws' from AOR 'adrian' due to shutdown
== Contact adrian/sip:pehgks63@86.18.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!

[david551]

You need to use RTP timeouts. Asterisk will continue transmitting media and that could well work, as,in general media end points can differ from the signalling ones.

[adrianhand]

Are you referring to rtp_timeout in my pjsip.conf endpoint, David?

[david551]

Probably. I’ve only used the similarly named option in chan_sip.