<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 href="mailto:sreenara@cisco.com" target="_blank">sreenara@cisco.com</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 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 href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>> </div>
<div>Cc: Cisco VoIP Group <<a 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></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></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 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">https://supportforums.cisco.com/blog/154506</a></p>
<p><span style="color:rgb(31,73,125)"></span></p>
<p> <br>
</p>
<p></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 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 href="mailto:nicksbarnett@gmail.com" target="_blank">nicksbarnett@gmail.com</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>
_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" rel="noreferrer" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div></div></div>
</blockquote></div><br></div>