[cisco-voip] Need SIP invite to route on TO field instead of URI

[Jason Burns]

IOS does have the ability to match on more than just the calling and called
numbers. Unfortunately the Cisco dial-peer matching document hasn't been
updated and I haven't found good official documentation, but look at this
link

https://supportforums.cisco.com/docs/DOC-25219

You can match on many parts of the SIP headers to route SIP calls. Even
more importantly, these header matches come before the usual calling and
called number type matches in terms of matching order. That doc should get
you started. It's for matching the incoming dial-peer, but the outbound
dial-peer configuration is similar.

Actually - here is something that is a bit closer, but still not perfect.
At Cisco Live I found some great slides describing this. Let me see if I
can find those slides and come back with a better answer.

http://www.cisco.com/en/US/docs/ios/voice/ivr/configuration/guide/gt_url.html#wp1064328

-Jason

On Mon, Jul 2, 2012 at 1:28 AM, Divin John <dijohn at cisco.com> wrote:

> Try SIP Profiles.
>
> http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sip/configuration/15-2mt/voi-condl-header.html#GUID-980E7543-A7EC-41F4-800A-0ECAFAD8899F
>
> From: Jonathan Charles <jonvoip at gmail.com>
> Date: Monday 2 July 2012 10:32 AM
> To: <cisco-voip at puck.nether.net>
> Subject: [cisco-voip] Need SIP invite to route on TO field instead of URI
>
> I have a CUBE running 15.1(4)M4 that is SIP to the provider and H.323 to
> CUCM 8.6.2
>
> Inbound calls are all showing the billing number on the request URI:
>
>
> Jul  1 09:48:43.709 CDT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> INVITE sip:1000 at 10.50.1.7:5060 SIP/2.0
> Via: SIP/2.0/UDP 1.1.1.1:5060
> ;branch=z9hG4bK-b09eefee3692d5c02b969912a045c958;rport
> From: "DOE, JOHN" <sip:16305551414 at call.message-alert.com>;tag=1813914414
> To: <sip:16305551212 at 1.1.1.1>
> Call-ID: 47e9ec2b at pbx
> CSeq: 22580 INVITE
> Max-Forwards: 70
> Contact: <sip:1000 at 1.1.1.1:5060;transport=udp>
> Supported: 100rel, replaces, norefersub
> Allow-Events: refer
> Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
> Accept: application/sdp
> User-Agent: talkingplatforms/2.1.15.2503
> Alert-Info: <Bellcore-dr3>
> Content-Type: application/sdp
> Content-Length: 323
>
> v=0
> o=- 1303493202 1303493202 IN IP4 66.159.89.13
> s=-
> c=IN IP4 1.1.1.1
> t=0 0
> m=audio 57094 RTP/AVP 0 9 18 2 3 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:18 g729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:2 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendrecv
>
>
>
>
> As you can see the INVITE is to sip:1000 at 10.50.1.7:5060, which is our
> billing code.
>
> The TO field has the actual DID to route to.
>
>
> Per the RFC, 3261, section 8.1.1.1:
>
> "The initial Request-URI of the message SHOULD be set to the value of the
> URI in the To field."
>
>
> This provider doesn't do that and says everyone but Cisco supports this,
> which goes back to my theory that SIP is not a protocol, but an idea of
> things you might want to do, but can really do whatever you want (hence the
> effect that every SIP provider seems to be doing their own thing)...
>
>
> Anyway, here is what I need to do:
>
> I need to have the call route on the TO field instead of the INVITE field.
>
> How?
>
>
>
> Jonathan
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