[OpenSIPS-Users] Handling missing BYEs

Callum Guy callum.guy at x-on.co.uk
Wed Mar 10 11:44:09 EST 2021


Might be a case of adding in a record_route() call in the appropriate
place, hard to say without a trace :)

On Wed, 10 Mar 2021 at 10:42, 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.
>
> Cheers,
>
> Mark
>
>
>
> 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?
>>
>> Callum
>>
>> 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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X-on
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