[cisco-voip] SIP Trunk on CallManager Gateway Router

Fri May 14 10:08:54 EDT 2010

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

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-----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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>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>>
>>>
>>>
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>>>
>>>
>>
>
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