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

[Jonathan Charles]

These are also for Outbound.... I need to match on an INBOUND To field.

On Mon, Jul 2, 2012 at 8:25 AM, Jason Burns <burns.jason at gmail.com> wrote:

> 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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>>
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>>
>
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