[OpenSIPS-Users] Handling missing BYEs

[Mark Allen]

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?
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