Nick, sorry to hear that. Maybe this release will get added to the compatibility matrix in the future after appropriate testing.

Dave, There's nothing wrong with this setup. It'll work.
However, if you're also using the router for SRST at a branch site, then you will need to force the phones to failover on the ITSP interface rather than the LAN voip interface.

Sreekanth

On Thursday 05 May 2016 03:23 AM, Dave Goodwin wrote:
Is there anything wrong with adding voice-class sip bind commands to ALL the voip dial-peers, and then set the global binding to the interface that faces the ITSP requiring authentication (since it seems sip-ua REGISTER messages use the global bind)?

-Dave

On Wed, May 4, 2016 at 4:22 PM, Nick Barnett <[email protected] <mailto:[email protected]>> wrote:

    Thanks for everybody's ideas.

    Unfortunately, 15.6 is OUT because it is not on the CVP 10.0
    compatibility matrix :(

    I'm going to look at using multiple registrars and see if I can
    trick it into behaving... if that doesn't work, I guess I'll have
    to remove my global binding...


    On Wed, May 4, 2016 at 11:35 AM, Sreekanth <[email protected]
    <mailto:[email protected]>> wrote:

        Yes, sip-ua tells CUBE to send REGISTER messages towards a
        Registrar server globally with the authentication and
        credential parameters. These REGISTER messages will be bound
        to the interface that is bound under voice service voip ->
        sip. However, in the 15.6(2)T version, the tenant
        configurations under the dial-peers will instruct the CUBE to
        send out REGISTER messages.

        I just checked with the router in my lab and actually, option
        2 won't be possible. It won't instruct the CUBE to send out
        REGISTER messages. It will only instruct the CUBE to add
        authentication credentials and realm settings when sending out
        the INVITE messages towards the session target configured
        under the dial-peer.

        You will have to go with option 1.
        *1. Create the voice class tenant for the SIP trunk to ITSP
        and bind it with the right interface.*
        voice class tenant 1
          registrar dns:cisco.com <http://cisco.com> expires 3600
          credentials username cisco password cisco realm cisco.com
        <http://cisco.com>
          authentication username cisco123 password 7 cisco123
          sip-server dns:cisco.com <http://cisco.com>
          bind control source-interface GigabitEthernet0/2
          bind media source-interface GigabitEthernet0/2
          early-offer forced

        *2. Apply the voice class tenant to the dial-peer. Create
        specific dial-peers towards ITSP.*
        dial-peer voice X voip
         voice-class sip tenant 1

        When this is done, CUBE will send REGISTER messages as well
        towards this ITSP with the traffic bound to gig0/2.
        This way you can have multiple ITSP trunks on 1 CUBE.

        Sreekanth




        On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
        I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can
        totally upgrade. Was actually planning on going to 15.4 this
        weekend. Jumping 3 versions kind of scares me, so maybe
        staging is in order.

        *I do have some limited auth commands on the dial peer, if
        this is what you were talking about... but I don't think it
        applies in this scenario. I don't have any options for
        credentials:*
        CUBE(config-dial-peer)#voice-class sip authenticate ?
          redirecting-number  Use redirecting number credentials
        while authenticating

        CUBE(config-dial-peer)#voice-class sip cred
        CUBE(config-dial-peer)#voice-class sip c?
          call-route  calltype-video  conn-reuse  copy-list
        *
        *
        *There is also the registration commands:*
        CUBE(config-dial-peer)#voice-class sip registration ?
          passthrough  SIP Registration Passthrough Options

        CUBE(config-dial-peer)#voice-class sip registration passthrough ?
          dynamic          SIP Registration Use dynamic Registrar
        Details (default)
          local-fallback   Local Fallback - (e2e)
          rate-limit       SIP Registration pass-through rate-limit
        Options
          reg-sync         Registration Sync - send REGISTER when
        registrar up (p2p)
          registrar-index  Registrar Index(s) used for registration
        passthrough
          static           SIP Registration Use static Registrar Details
          system           Use global registration passthrough CLI
        setting
          <cr>

        *I tried using the system passthrough setting, but it did not
        work.*
        *
        *
        *I need to make sure I understand what is actually happening.*
        *
        *
        *I don't think the CUBE is even looking at dial-peers for
        REGISTER messages. Am I correct?  If so, no amount of dial
        peer settings is going to make any difference here... unless
        there is a way to create a dial-peer that will intercept
        REGISTER messages. I believe it is using the REALM settings
        in the credentials and authentication strings (that I entered
        into sip-ua). And sip-ua is using the global bind settings I
        set within voice service voip -> SIP (which are set to the
        inside interface).*
        *
        *
        *Please set me straight!*

        Thanks,
        Nick

        On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan
        (sreenara) <[email protected] <mailto:[email protected]>>
        wrote:

            What IOS version are you running on the CUBE? I can think
            of a couple of things.
            1. In 15.6(2)T, a new feature has been introduced called
            multi-tenant where you can configure separate voice class
            tenants. Each tenant can have separate authentication
            mutually exclusive to one another and can be bound to
            different interfaces.

            2. In your current IOS, check if you are able to
            configure the authentication and credential commands at
            the dial peer level. I am not sure which IOS had this
            introduced but it is worth a try.



            Sreekanth

            Sent from a phone.


            -------- Original message --------
            From: Nick Barnett <[email protected]
            <mailto:[email protected]>>
            Date: 5/4/16 8:03 PM (GMT+05:30)
            To: Brian Meade <[email protected] <mailto:[email protected]>>
            Cc: Cisco VoIP Group <[email protected]
            <mailto:[email protected]>>
            Subject: Re: [cisco-voip] Authenticating sip trunk to
            ITSP from CUBE?

            I'm binding control and media to my inside interface:

            sip

              bind control source-interface GigabitEthernet0/0
              bind media source-interface GigabitEthernet0/0

            I suspect this is the issue... is there any way to make
            the REGISTER messages come from the outside gi0/1 interface?

            The reason I'm binding to inside is that we have a a very
            fluid internal network. I have to make and modify
            internal dial peers almost daily.  When I need to create
            a dial peer and put the bind statements on the dial peer,
            it won't bind properly since there are active SIP calls
            on the CUBE... so I bound it globally. My external dial
            peers rarely change, so I bind those directly to gi0/1
            (on the DP).

            I was under the impression that REGISTER events can take
            place without a dial peer... but is there a way to, i
            dunno, make a dial peer for register messages? Can I use
            SIP profile magic to get it working as is?

            I found this article which is pretty much exactly what
            I'm dealing with, but it doesn't mention REGISTER at all...

            https://supportforums.cisco.com/blog/154506



            On Wed, May 4, 2016 at 9:06 AM, Brian Meade
            <[email protected] <mailto:[email protected]>> wrote:

                Do you already have the SIP bind under voice service
                voip?
                voice service voice
                 sip
                  bind all source-interface FastEthernet0

                On Wed, May 4, 2016 at 9:58 AM, Nick Barnett
                <[email protected]
                <mailto:[email protected]>> wrote:

                    I've never dealt with an authenticated SIP trunk
                    before and I'm having some issues. I was
                    wondering if anyone has had a similar experience.
                    I already have 2 SIP trunks from ITSP-1 that do
                    NOT require authentication. These are working
                    fine and have been for years.

                    We are adding ITSP-2 and their SIP service DOES
                    require auth. I've followed their integration
                    guide (which left a lot to be desired) and their
                    acceptance team is telling me my auth is coming
                    from our private class A address.

                    Our CUBE is in HA with an inside (10.x.x.x) and
                    outside (public) IP address. They are seeing
                    REGISTER messages sourcing the inside VIP.

                    I was looking around for an auth BIND statement
                    or something like that, but I haven't had any
                    luck. Any pointers?

                    Thanks,
                    Nick

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