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

Mon Jul 2 18:26:56 EDT 2012

Well, all of the solutions are for SIP to SIP only.... I can try it...

On Mon, Jul 2, 2012 at 4:51 PM, Jason Burns <burns.jason at gmail.com> wrote:

> That's certainly an interesting problem.
>
> You have an inbound SIP message, and selecting the inbound dial-peer isn't
> really what you're concerned with. You're more concerned that the outbound
> H.323 dial-peer selection is only going to look at the called number in the
> SIP Request URI (which is the billing number).
>
> It sounds like you need to do some work on the inbound SIP dial-peer to
> manipulate that Request URI and stick the TO header inside of it. Then you
> want to have this message processed by IOS  to find the right H.323
> dial-peer based on your modification.
>
> Another option is to change the header that IOS looks at when going from
> SIP to H323 for pulling out the called number, but I have no idea how to do
> that.
>
> This example would copy the TO header into the Request-URI portion, but I
> think you're right that it's really only meant for SIP to SIP. We'd pass
> the modified SIP message out the far side, but I don't know if we'd make
> the modification on the inbound leg and then change our next hop routing
> based on this modification. I don't think it was intended for SIP to H.323.
>
> https://supportforums.cisco.com/thread/2119596
>
>
> How about converting the CUBE to be SIP-SIP? There may be other solutions
> but I'm not brushed up enough on my IOS SIP modifications to come up with
> something better! Anyone else have ideas?
>
> -Jason
>
>
>
>
> On Mon, Jul 2, 2012 at 10:12 AM, Jonathan Charles <jonvoip at gmail.com>wrote:
>
>> 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
>>>> _______________________________________________ cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>> _______________________________________________
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>>>>
>>>>
>>>
>>
>
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