[OpenSIPS-Users] Handling missing BYEs
mark at allenclan.co.uk
Wed Mar 10 16:08:18 EST 2021
Just a follow-up for fix notes. We've got OpenSIPS sitting behind a
firewall with a 1:1 port map from an external IP. The Contact HF for the
INVITE going out from OpenSIPS to the NATed UAC used the internal IP
address for the OpenSIPS box, not the external IP. Set OpenSIPS to fix it
and all working fine now. Thanks for the help
On Wed, 10 Mar 2021 at 10:39, Mark Allen <mark at allenclan.co.uk> wrote:
> Hi Callum - thanks for that!
> Yes - it's generating the BYE at the Linux end but not sending it to the
> remote OpenSIPS IP address but rather to an address on the local LAN -
> hence the problem. Thanks for your help.
> On Wed, 10 Mar 2021 at 09:26, Callum Guy <callum.guy at x-on.co.uk> wrote:
>> 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
>>> 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
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