[cisco-voip] SIP Trunk on CallManager Gateway Router

[O'Brien, Neil]

The CUE dial peer is hardcoded to g711.  I'm not sure what the calls are coming in on though.  There's no one there at the moment so I can't get a call connected to see what the codec is.

There is transcending configured but it's associated with the call manager server - does that matter?

Regards,
 
Neil O'Brien
Network Engineer

Datapac Ltd
Phone: +353 1 426 3500
Fax:    +353 1 426 3501
Email:  nobrien at datapac.com
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-----Original Message-----
From: ccieid1ot [mailto:ccieid1ot at gmail.com] 
Sent: 13 May 2010 22:58
To: O'Brien, Neil
Cc: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router

Is the dial-peer to the CUE hard coded to G.711?  Also, you will need
a transcoder if calls from the SIP trunk is G.729.

On Thu, May 13, 2010 at 4:48 PM, O'Brien, Neil <nobrien at datapac.com> wrote:
> That was the money shot Ryan, it registered once I configured a pots
> dial-peer.
>
>
>
> I'm now having trouble getting the SIP trunk to talk to CUE, I think it
> might be a codec mismatch.
>
>
>
>
>
>
>
> 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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