[OpenSIPS-Users] check_source_address()

[Mark Farmer]

Thanks Ben, that seems to be the case. I will update the list with the
results.

Mark.


On Fri, 26 Apr 2019 at 14:09, Ben Newlin <Ben.Newlin at genesys.com> wrote:

> It would cause the issue if they are sending all requests to that domain,
> including sequential requests like re-invite, and ignoring the Contact
> provided in the 200 OK. That is not correct according to RFC 3261, but I
> have seen many carriers do this.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Mark Farmer
> <farmorg at gmail.com>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Friday, April 26, 2019 at 8:59 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] check_source_address()
>
>
>
> Thank you, that makes sense now. I will keep that in mind for the future.
>
> In the meantime I have raised a query with our provider.
>
>
>
> Additionally, I realised this morning that at our request, our provider is
> sending calls to us via a domain name instead of an IP. Would that likely
> cause the issue even if they are using RFC 3261? I have asked for it to be
> removed.
>
>
>
> Best regards
>
> Mark.
>
>
>
>
>
>
>
> On Thu, 25 Apr 2019 at 16:50, Liviu Chircu <liviu at opensips.org> wrote:
>
> On 25.04.2019 17:11, Mark Farmer wrote:
>
> Thanks so much for helping with this.
>
>
>
> I have applied the suggested config but the result is the same. OpenSIPS
> routes the RE-INVITE to itself and it never gets routed back to the
> Asterisk box.
>
> If the 2nd Route header in the RE-INVITE is the IP of the other interface
> - will that not always be the case? It's as though the 2nd Route header
> needs to be changed to have the IP of the Asterisk server.
>
>
>
> Sanitized RE-INVITE from provider:
>
>
>
> INVITE sip:asterisk at my.host.name:5060 SIP/2.0
>
>
>
> If OpenSIPS identifies "my.host.name:5060" as a local domain, this will
> screw up the routing,
> as it will go from loose (RFC 3261) to strict (old, deprecated RFC 2543
> mechanism).  Notice how
> its not preserving the R-URI when it routes to itself as should happen
> with RFC 3261 routing,
> because it has fallen back to RFC 2543 routing.
>
> Your provider needs to follow RFC 3261 and use as Re-INVITE Request-URI
> the exact Contact
> advertised by the caller: <sip:asterisk at 10.98.0.102:5060>
> <sip:asterisk at 10.98.0.102:5060>, and not confuse your routing engine
> with a random target such as: INVITE sip:asterisk at my.host.name:5060.
>
> --
>
> Liviu Chircu
>
> OpenSIPS Developer
>
> http://www.opensips-solutions.com
>
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>
>
>
>
> --
>
> Mark Farmer
> farmorg at gmail.com
> _______________________________________________
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>


-- 
Mark Farmer
farmorg at gmail.com
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