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-boun...@lists.opensips.org> on behalf of Mark Farmer <farm...@gmail.com> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Friday, April 26, 2019 at 8:59 AM To: OpenSIPS users mailling list <users@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 <li...@opensips.org<mailto:li...@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:aster...@my.host.name:5060<http://sip:aster...@my.host.name:5060> SIP/2.0 If OpenSIPS identifies "my.host.name:5060<http://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@10.98.0.102:5060><mailto:sip:asterisk@10.98.0.102:5060>, and not confuse your routing engine with a random target such as: INVITE sip:aster...@my.host.name:5060<mailto:sip:aster...@my.host.name:5060>. -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com<http://www.opensips-solutions.com> _______________________________________________ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> -- Mark Farmer farm...@gmail.com<mailto:farm...@gmail.com>
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