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    <font size="-1"><font face="Helvetica, Arial, sans-serif">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.<br>
        <br>
        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.<br>
        <br>
        You will have to go with option 1.<br>
        <b>1. Create the voice class tenant for the SIP trunk to ITSP
          and bind it with the right interface.</b><br>
        voice class tenant 1<br>
          registrar dns:cisco.com expires 3600<br>
          credentials username cisco password cisco realm cisco.com<br>
          authentication username cisco123 password 7 cisco123<br>
          sip-server dns:cisco.com<br>
          bind control source-interface GigabitEthernet0/2<br>
          bind media source-interface GigabitEthernet0/2<br>
          early-offer forced<br>
        <br>
        <b>2. Apply the voice class tenant to the dial-peer. Create
          specific dial-peers towards ITSP.</b><br>
        dial-peer voice X voip<br>
         voice-class sip tenant 1<br>
        <br>
        When this is done, CUBE will send REGISTER messages as well
        towards this ITSP with the traffic bound to gig0/2.<br>
        This way you can have multiple ITSP trunks on 1 CUBE.<br>
        <br>
        Sreekanth<br>
        <br>
        <br>
        <br>
      </font></font><br>
    <div class="moz-cite-prefix">On Wednesday 04 May 2016 09:29 PM, Nick
      Barnett wrote:<br>
    </div>
    <blockquote
cite="mid:CAPtKLNDv3mJZCuLDDs0qnbwFUwNoAApy5Rk6=1kodVt79aJHMA@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html; charset=utf-8">
      <div dir="ltr">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.
        <div><br>
        </div>
        <div><b>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:</b></div>
        <div>CUBE(config-dial-peer)#voice-class sip authenticate ?<br>
        </div>
        <div>
          <div>  redirecting-number  Use redirecting number credentials
            while authenticating</div>
          <div><br>
          </div>
          <div>CUBE(config-dial-peer)#voice-class sip cred         </div>
          <div>CUBE(config-dial-peer)#voice-class sip c?  </div>
          <div>  call-route  calltype-video  conn-reuse  copy-list</div>
          <div><b><br>
            </b></div>
          <div><b>There is also the registration commands:</b></div>
          <div>
            <div>CUBE(config-dial-peer)#voice-class sip registration ?<br>
            </div>
            <div>  passthrough  SIP Registration Passthrough Options</div>
            <div><br>
            </div>
            <div>CUBE(config-dial-peer)#voice-class sip registration
              passthrough ?<br>
            </div>
            <div>  dynamic          SIP Registration Use dynamic
              Registrar Details (default)</div>
            <div>  local-fallback   Local Fallback - (e2e)</div>
            <div>  rate-limit       SIP Registration pass-through
              rate-limit Options</div>
            <div>  reg-sync         Registration Sync - send REGISTER
              when registrar up (p2p)</div>
            <div>  registrar-index  Registrar Index(s) used for
              registration passthrough</div>
            <div>  static           SIP Registration Use static
              Registrar Details</div>
            <div>  system           Use global registration passthrough
              CLI setting</div>
            <div>  <cr></div>
          </div>
          <div><br>
          </div>
          <div><b>I tried using the system passthrough setting, but it
              did not work.</b></div>
        </div>
        <div><b><br>
          </b></div>
        <div><b>I need to make sure I understand what is actually
            happening.</b></div>
        <div><b><br>
          </b></div>
        <div><b>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).</b></div>
        <div><b><br>
          </b></div>
        <div><b>Please set me straight!</b></div>
        <div><br>
        </div>
        <div>Thanks,</div>
        <div>Nick</div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Wed, May 4, 2016 at 10:37 AM,
          Sreekanth Narayanan (sreenara) <span dir="ltr"><<a
              moz-do-not-send="true" href="mailto:sreenara@cisco.com"
              target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:sreenara@cisco.com">sreenara@cisco.com</a></a>></span> wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div>
              <div>What IOS version are you running on the CUBE? I can
                think of a couple of things.</div>
              <div>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.</div>
              <div><br>
              </div>
              <div>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.</div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div>
                <div style="font-size:85%;color:#575757">Sreekanth</div>
                <div style="font-size:85%;color:#575757"><br>
                </div>
                <div style="font-size:85%;color:#575757">Sent from a
                  phone.</div>
              </div>
              <span class="">
                <div><br>
                </div>
                <div><br>
                </div>
                <div>-------- Original message --------</div>
                <div>From: Nick Barnett <<a moz-do-not-send="true"
                    href="mailto:nicksbarnett@gmail.com" target="_blank">nicksbarnett@gmail.com</a>>
                </div>
                <div>Date: 5/4/16 8:03 PM (GMT+05:30) </div>
                <div>To: Brian Meade <<a moz-do-not-send="true"
                    href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>>
                </div>
                <div>Cc: Cisco VoIP Group <<a moz-do-not-send="true"
                    href="mailto:cisco-voip@puck.nether.net"
                    target="_blank">cisco-voip@puck.nether.net</a>> </div>
                <div>Subject: Re: [cisco-voip] Authenticating sip trunk
                  to ITSP from CUBE? </div>
                <div><br>
                </div>
              </span>
              <div>
                <div class="h5">
                  <div>
                    <div dir="ltr">
                      <p>I'm binding control and media to my inside
                        interface:</p>
                      <p>sip      </p>
                      <p>  bind control source-interface
                        GigabitEthernet0/0<br>
                          bind media source-interface GigabitEthernet0/0<br>
                      </p>
                      <p>I suspect this is the issue... is there any way
                        to make the REGISTER messages come from the
                        outside gi0/1 interface?</p>
                      <p>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).<br>
                      </p>
                      <p>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?<br>
                      </p>
                      <p>I found this article which is pretty much
                        exactly what I'm dealing with, but it doesn't
                        mention REGISTER at all...</p>
                      <p><span
                          style="font-family:Symbol;color:rgb(31,73,125)"><span
                            style="font-size:7pt;font-family:'Times New
                            Roman'">   </span></span><a
                          moz-do-not-send="true"
href="https://urldefense.proofpoint.com/v2/url?u=https-3A__supportforums.cisco.com_blog_154506&d=CwMFAg&c=M-KQspD_LQogCbR-BWCHOaeDEPOhF8vWqHZTaiwxT3c&r=T9uVLZucbHG2NKKKzOrp-o5cpdReHj02PkJJsCVkgfwcv7S0R5lDeFJg2VRbiNih&m=UIAzGDQs8RCZld9kCbExwqpJhTgzpDVwM0k8_I7JRqU&s=jZN-R2pRsZOWN3r5is-aSivDlf9hqddUzDIoOWRWc3E&e="
                          target="_blank"><a class="moz-txt-link-freetext" href="https://supportforums.cisco.com/blog/154506">https://supportforums.cisco.com/blog/154506</a></a></p>
                      <p><span style="color:rgb(31,73,125)"></span></p>
                      <p> <br>
                      </p>
                    </div>
                    <div class="gmail_extra"><br>
                      <div class="gmail_quote">On Wed, May 4, 2016 at
                        9:06 AM, Brian Meade <span dir="ltr">
                          <<a moz-do-not-send="true"
                            href="mailto:bmeade90@vt.edu"
                            target="_blank">bmeade90@vt.edu</a>></span>
                        wrote:<br>
                        <blockquote class="gmail_quote" style="margin:0
                          0 0 .8ex;border-left:1px #ccc
                          solid;padding-left:1ex">
                          <div dir="ltr">Do you already have the SIP
                            bind under voice service voip?
                            <div>voice service voice</div>
                            <div> sip</div>
                            <div>  bind all source-interface
                              FastEthernet0</div>
                          </div>
                          <div class="gmail_extra"><br>
                            <div class="gmail_quote">
                              <div>
                                <div>On Wed, May 4, 2016 at 9:58 AM,
                                  Nick Barnett <span dir="ltr"><<a
                                      moz-do-not-send="true"
                                      href="mailto:nicksbarnett@gmail.com"
                                      target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:nicksbarnett@gmail.com">nicksbarnett@gmail.com</a></a>></span>
                                  wrote:<br>
                                </div>
                              </div>
                              <blockquote class="gmail_quote"
                                style="margin:0 0 0 .8ex;border-left:1px
                                #ccc solid;padding-left:1ex">
                                <div>
                                  <div>
                                    <div dir="ltr">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.
                                      <div><br>
                                      </div>
                                      <div>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.
                                        <div><br>
                                        </div>
                                        <div>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.</div>
                                        <div><br>
                                        </div>
                                        <div>I was looking around for an
                                          auth BIND statement or
                                          something like that, but I
                                          haven't had any luck. Any
                                          pointers?</div>
                                        <div><br>
                                        </div>
                                        <div>Thanks,</div>
                                        <div>Nick</div>
                                      </div>
                                    </div>
                                    <br>
                                  </div>
                                </div>
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                                <a moz-do-not-send="true"
                                  href="https://puck.nether.net/mailman/listinfo/cisco-voip"
                                  rel="noreferrer" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
                                <br>
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