[cisco-voip] SIP Trunk on CallManager Gateway Router

[ccieid1ot]

Is the CUE registered to CUCM?  You can hardcode the dial-peer coming
in from the SIP provider to G.711 and see if that works.

On Thu, May 13, 2010 at 5:07 PM, O'Brien, Neil <nobrien at datapac.com> wrote:
> 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
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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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>>
>>
>>
>>
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