[OpenSIPS-Users] Inbound DID TO/INVITE issue

[Brett Nemeroff]

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 <mike at oeg.com.au> 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:111610 at 150.10.10.10:49640 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:0882229300 at 202.13.13.13:5060>
> To: <sip:0883622272 at 202.11.11.11;user=phone>
> From: "0882229300"<sip:0882229300 at 202.13.13.13;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:0883622272 at 150.10.10.10:49640 SIP/2.0
>
> Any idea on how I might correct this ?
> What should the above INVITE really look like ?
>
> Thanks
> Mike
>
>
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