[cisco-voip] SIP Trunk on CallManager Gateway Router

[ccieid1ot]

Do a debug ccsip all.

On Fri, May 14, 2010 at 9:08 AM, O'Brien, Neil <nobrien at datapac.com> wrote:
> Thanks nick - it's there alright.  I forced both the SIP trunk and the CUE dial-peer to G711ulaw and still no joy.
>
> I'm not even sure of what debugs I need to run to see what's going on?
>
> Regards,
>
> Neil O'Brien
> Network Engineer
>
> Datapac Ltd
> Phone: +353 1 426 3500
> Fax:    +353 1 426 3501
> Email:  nobrien at datapac.com
> Web:   www.datapac.com
>
>
> PREMIER PARTNER OF THE YEAR 2009
>
> Datapac is the leading Irish business technologies provider
> IT Maintenance & Managed Services | Virtualisation & Storage Solutions | Microsoft Infrastructure Solutions
> Voice & Data Networks | Unified Communications | Microsoft Dynamics Nav ERP | EPOS Retail Solutions | IT Consumables
>
>
> -----Original Message-----
> From: matthn at gmail.com [mailto:matthn at gmail.com] On Behalf Of Nick Matthews
> Sent: 14 May 2010 15:02
> To: ccieid1ot
> Cc: O'Brien, Neil; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>
> Make sure you have
>
> voice service voip
>  allow-connections sip-to-sip
>
> -nick
>
> On Thu, May 13, 2010 at 6:24 PM, ccieid1ot <ccieid1ot at gmail.com> wrote:
>> 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
>>> Fax:    +353 1 426 3501
>>> Email:  nobrien at datapac.com
>>> Web:   www.datapac.com
>>>
>>>
>>> PREMIER PARTNER OF THE YEAR 2009
>>>
>>> Datapac is the leading Irish business technologies provider
>>> IT Maintenance & Managed Services | Virtualisation & Storage Solutions | Microsoft Infrastructure Solutions
>>> Voice & Data Networks | Unified Communications | Microsoft Dynamics Nav ERP | EPOS Retail Solutions | IT Consumables
>>>
>>>
>>> -----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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>>>>
>>>
>>
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