[OpenSIPS-Users] Handling missing BYEs
Johan De Clercq
Johan at democon.be
Wed Mar 10 12:21:09 EST 2021
@Callum Guy <callum.guy at x-on.co.uk> indeed :-)
Op wo 10 mrt. 2021 om 12:47 schreef Callum Guy <callum.guy at x-on.co.uk>:
> 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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>> _______________________________________________
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>
>
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