[cisco-voip] SIP Trunk on CallManager Gateway Router

[O'Brien, Neil]

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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