Hi Peter,

According to RFC3261, in SIP, routing is exclusively done based on RURI (and eventually Route hdrs). TO header has 0 implication in routing. So, your PBX is deprecated and out of standards and I suggest either upgrade, either replacement.

Regards, Bogdan

Peter P GMX wrote:

I have the following scenario:

I receive a request with PSTN_number@my.ip.add.res from my VoIP In alias_db I have defined a rule that points to yyy@myip.com which is a registered account

After alias_db_lookup OpenSER invites as follows: Invite yyy@myip.com To: xxxxxxx@my.ip.add.res

The UA (actually a PBX with VoIP support) answers with 404 not found.

When I invite as follows: Invite yyy@myip.com To: yyy@myip.com

everything works fine.

My question: How do you handle these cases? Rewriting the To: part seems dificult and will mess up a lot - hein?

Kind regards Peter