[OpenSIPS-Users] Handling missing BYEs

Callum Guy callum.guy at x-on.co.uk
Wed Mar 10 09:23:50 EST 2021

Hi Mark,

It sounds like you may be having issues with the proxy not keeping itself
in path for certain call scenarios.

Are you able to provide a SIP trace and/or opensips config? Also if you're
running Blink on a Linux system, can you get a SIP trace there to see if
the BYE is being generated and sent somewhere else?


On Tue, 9 Mar 2021 at 16:32, Mark Allen <mark at allenclan.co.uk> wrote:

> I'm seeing some odd behaviour which also leads into a broader question
> I have a NATed Blink app running on Linux on my home LAN. It connects to
> an OpenSIPS 3.1 server in on our office LAN which is a mid-registrar for an
> Asterisk server. I'm running sngrep on the OpenSIPS box to watch the
> traffic.
> If I call from the Blink app to another extension it all connects and
> audio works correctly. If I hangup in Blink, a BYE is sent via OpenSIPS to
> Asterisk - all good so far.
> If I call from another extension to the Blink app it all connects and
> audio works correctly. However, if I hangup in the Blink app, no BYE is
> sent to OpenSIPS.
> In most situations, this is merely inconvenient because, with the loss of
> RTP traffic, Asterisk generates a BYE after about 35 seconds to tidy
> everything up. However, if I'm doing an attended transfer, the BYE is
> needed to exit the call so that the transfer completes successfully. At the
> moment, if I hangup in the Blink app, there's a wait of 35 seconds until
> Asterisk creates the BYE before the call transfer is completed.
> While I'm mostly using Blink, I've seen similar failures to send BYEs from
> other apps. Does OpenSIPS offer anything that could help with this?
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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