[OpenSIPS-Users] Inbound DID TO/INVITE issue

[Mike O'Connor]

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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