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

[Jonathan Charles]

This only applies to an outgoing message, not the incoming.... I have tried
this and only works if you are SIP to SIP, we are not (we also terminate a
PRI on the router).

On Mon, Jul 2, 2012 at 12:28 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
> _______________________________________________ cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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