Well hold on a sec.. First of all, the TO field is irrelevant. So whatever RURI you have (that's the top line INVITE URI), that's where we're sending the call to next. If the below invite hits asterisk it should be delivered to 111610. If that's not right, you need to set your $rU to whatever you want it to be delivered to.
Per the docs, the function you are using updates the RURI: http://www.opensips.org/html/docs/modules/1.7.x/alias_db.html#id250076 Are you suggesting asterisk is routing on the TO header? This happens with some buggy SIP clients from time to time, but I wouldn't expect this in Asterisk. The "To" Header really shouldn't be considered for routing. That being said, there are a handful of UAs out there that insist on doing so. I think they are pre-3261 typically but this isn't confirmed. -Brett On Tue, Oct 16, 2012 at 11:05 PM, Mike O'Connor <[email protected]> wrote: > Hi All > > I've had a system setup for a long time, but one issue has always been > there and its come to a head. > > I've always has problems with Asterisk not correctly selecting the call > route for inbound DID's because the INVITE sent to it via my core > (openSIPs) has the 'service number' not the DID in the INVITE. > > As per below > > INVITE sip:[email protected]:49640 SIP/2.0 > Record-Route: > <sip:202.11.11.11;lr=on;ftag=1ca96a43-co774-INS001;did=781.c6890334> > Via: SIP/2.0/UDP 202.11.11.11;branch=z9hG4bK2ad4.58ca31d3.0 > Via: SIP/2.0/UDP > 202.13.13.13:5060;branch=z9hG4bK1ca53b72111cfdd3INV1ca96a43306 > Max-Forwards: 34 > Contact: <sip:[email protected]:5060> > To: <sip:[email protected];user=phone> > From: > "0882229300"<sip:[email protected];user=phone;noa=national>;tag=1ca96a43-co774-INS001 > Call-ID: 6f75-572-026197035530-IMG01-0-27.34.224.68 > CSeq: 77401 INVITE > Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO > Content-Type: application/sdp > Date: Wed, 17 Oct 2012 03:09:42 GMT > User-Agent: ENSR3.0.66.21-IS1-RMRG109-RG2100-CPO46 > Content-Length: 309 > > On an inbound call from our wholesale supplier I run a command > alias_db_lookup which I think changes the DID the service number. > > if (is_method("INVITE")) { > ... > ... > ... > if(alias_db_lookup("dbaliases","d")) { > ... > ... > } > > I'm moved to supplying a commercially support Asterisk install for our > customers and they (the commercial asterisk company) are saying that this > rewrite is not correct and the INVITE should look like this. > > INVITE sip:[email protected]:49640 SIP/2.0 > > Any idea on how I might correct this ? > What should the above INVITE really look like ? > > Thanks > Mike > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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