[cisco-voip] SIP Trunk on CallManager Gateway Router

Fri May 28 11:22:19 EDT 2010

This sounds like a firewall / routing issue.  I would check to make
sure a few things:

-If there is a firewall involved it is SIP aware and configured
-Make sure your CUBE is bound to the right address.  It's very
possible the provider is sending to one of CUBE's addresses, but CUBE
is using a different address to respond that it's not happy with.

-nick

On Fri, May 28, 2010 at 9:46 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
> Not necessarily, but it will introduce a bit of complexity as the router has
> to do the h.323 to sip internetworking.
> In this case you should see CUCM send an H.225 Connect message to the
> router, which should generate a SIP 200 OK out to the provider, but possibly
> not until the router has completed H.245 media negotiations with CUCM.
> If what I'm saying doesn't make sense it's time to engage TAC.
> -Ryan
> On May 28, 2010, at 9:38 AM, O'Brien, Neil wrote:
>
> Thanks Ryan,
>
> But the CUCM isn’t talking SIP at all though, the gateway is still a H323
> gateway.  Is this my problem??
>
> Thanks,
> Neil
>
>
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: 28 May 2010 14:21
> To: O'Brien, Neil
> Cc: Frank Arrasmith; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>
> It may be mostly semantics but at the point your router is doing IP to IP
> it's acting as a CUBE.  Traditional voice gateways are IP to POTS.
>
> I'd disagree that your signaling is good however.  If the phone answers, and
> that is not communicated back out the SIP trunk then something is not right
> in the signaling.
>
> You need to look at the SIP messaging on the CUCM and at the router, just to
> make sure everything CUCM sends is passed on.
>
> -Ryan
>
> On May 27, 2010, at 5:01 PM, O'Brien, Neil wrote:
>
> Hi Frank – it’s not a CUBE router.  Forgive my ignorance but should it be?
>
> Thanks,
>
> Neil
>
> From: Frank Arrasmith [mailto:frank.arrasmith at gmail.com]
> Sent: 27 May 2010 18:40
> To: O'Brien, Neil
> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>
>
> This is just a shot, but it sounds like the signaling is good, so maybe
> something with the media? Is your CUBE set for flow-through, or
> flow-around?  When I set mine up, I had 2 problems, I messed with the media
> settings, and changed to flow-around, which my equipment didn't support. It
> should be flow-through, which I think is default.  The other problem I had
> was with binding the media to an interface.  I either had problems with
> routing to the interface, or bound the media to the wrong interface, I can't
> remember off the top of my head.  Just a couple more areas to look
> at...Please post back if you find a solution, as SIP/CUBE issues seem to pop
> up more and more these days.
>
> --Frank
>
> On Thu, May 27, 2010 at 4:15 AM, O'Brien, Neil <nobrien at datapac.com> wrote:
> Hi Guys,
>
> Dragging this one up again unfortunately!!
>
> So when the phones register back to CCM, I have a dummy pots dial-peer (as
> Ryan suggested) created so the SIP trunk remains registered.
>
> So at this point, all phones are registered to CCM.  I call in on the SIP
> trunk, the phone rings, when I answer nothing happens and the caller
> continues to hear ringback, then both disconnect.
>
> Logically, the way I see it is that the SIP call comes in on the IOS gateway
> and hits the incoming SIP dial-peer, it gets bumped over to the CCM via
> another dial-peer and CCM rings the phone in question.  This is all
> signalling.  What should then happen is CCM connects the phone and the SIP
> call and then drops out of the loop so the phone is talking directly with
> it’s local gateway that terminates the sip trunk.
>
> Somewhere, this is failing and I’ve no idea where to look at this point so
> any help is appreciated.
>
> Thanks,
>
> Neil
>
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: 13 May 2010 21:28
> To: O'Brien, Neil
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>
> Not my area of expertise, but a colleague pointed out that in order for the
> router to register with the sip provider it needs a pots dial-peer.  This
> happens automatically when the phones register via SRST.
>
> Try creating a dummy pots dial-peer and see if that works.
>
> -Ryan
>
> On May 13, 2010, at 4:20 PM, O'Brien, Neil wrote:
>
>
>
> Thanks ryan, but when the phones register with CM (ie. Come out of SRST)
> incoming SIP calls don’t even hit the router and the provider sees it as not
> registered.
>
> The router only seems to register with the provider when the phones register
> with the router (ie. Go into srst)
>
> Am I missing something?
>
>
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: 13 May 2010 21:12
> To: O'Brien, Neil
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>
> In CCM 4.x nothing should be talking SIP to CCM.  You should be just fine
> doing H.323 to the gateway and SIP out from there.   This is referred to as
> a CUBE, or IP-IP gateway setup.
>
> If the calls aren't even hitting the router as H.323 ('debug h225 asn1'
> doesn't show anything) then you need to look at the CCM config.  If the call
> is reaching the router but not going out SIP then check to make sure you
> have enabled the appropriate 'allow connection' commands under 'voice
> service voip'.
>
> -Ryan
>
> On May 13, 2010, at 3:32 PM, O'Brien, Neil wrote:
>
>
> Hi,
>
> A customer of mine has Call Manager 4.x and a few remote sites.  One of the
> sites was connected via an MPLS network and had a PSTN connection on their
> local gateway router.  They moved premises and ditched the MPLS connection
> and instead of getting a new PSTN line, they went down the SIP road and got
> a SIP trunk to an internet provider.  All the phones continued to work via
> SRST to the local router and the router was configured with the SIP trunk
> details as if it were a Call Manager Express setup.
>
> The remote site now has a VPN back to the HQ and the phones are now
> registering back with Call Manager.  Their local gateway router is still
> configured as that site’s Call Manager H323 gateway and I thought that once
> Call Manager pushed the calls to the gateway the call would continue to go
> out over the SIP trunk.
>
> Unfortunately, the SIP trunk only seems to register with the provider when
> the phones register in SRST on the router.  When the phones register back to
> call manager, the router will not register with the SIP provider.  Outbound
> calls will still go out over the SIP but incoming calls will not work.  I’m
> told by the provider that if the router isn’t registered with the SIP
> provider, outbound calls will still go over it but inbound calls will not.
>
> It’s not a dial peer routing issue as when I debug the dial peers while the
> phones are registered to Call Manager and make an inbound call I don’t see
> anything hit the router.  When the register in SRST, I see the SIP calls
> come in and match the relevant dial peers.
>
> I’m really at a loss as to why this would be, I don’t know very well how SIP
> works with Call Manager.  I’d really appreciate some pointers on this...
>
> Thanks for reading!!
>
> Neil
>
>
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