[asterisk-dev] SIP Peer matching on Inbound calls (and a featurerequest, please)

[Steve Langstaff]

> > -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com 
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of 
> Daniel Corbe
> Sent: 10 November 2006 21:05
> To: Asterisk Developers Mailing List
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-dev] SIP Peer matching on Inbound calls 
> (and a featurerequest, please)

> <-- SIP read from 64.49.129.5:5061:
> INVITE sip:441612412070 at 12.109.47.235:5060 SIP/2.0
> Via: SIP/2.0/UDP
> 64.49.129.5:5061;branch=z9hG4bK961d25b39e3acb6b937d9bce549c4d33;rport
> Max-Forwards: 70
> From: <sip:None at 64.49.129.5>;tag=9a658fc7987c766e4421588ccab07729
> To: <sip:441612412070 at 12.109.47.235>
> Digium's paid support is telling me that the only matching 
> criteria is either registered peers (and since this is a DID, 
> there are none) or the HOST to which the traffic is coming from.

[SNIP]
 
> I would like to see a 3rd match criteria here, and that is 
> for unregistered peers (as in the case of a DID) the Username 
> in the To:
> header needs to be matchable to a SIP peer.

I *think* that you will probably want to match the URI in the INVITE
line, rather than the one in the To: header. In this example the field
is more-or-less the same, but they can be different, and it's the URI in
the INVITE that identifies the resource being requested *at this
endpoint*.