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 expires 3600
credentials username cisco password cisco realm cisco.com
authentication username cisco123 password 7 cisco123
sip-server dns: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
<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=>
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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