[cisco-voip] SIP Trunk on CallManager Gateway Router

Fri May 28 14:12:30 EDT 2010

Can you also make sure
1) wait for farend h245 caps is unchecked on h323 trunk in ccm
2) emptycapability string and h225 nonstandard tcspassthrough enabled  
under voice service voip/ h323

Sent from my phone

On May 28, 2010, at 10:22 AM, Nick Matthews <matthnick at gmail.com> wrote:

> This sounds like a firewall / routing issue.  I would check to make
> sure a few things:
>
> -If there is a firewall involved it is SIP aware and configured
> -Make sure your CUBE is bound to the right address.  It's very
> possible the provider is sending to one of CUBE's addresses, but CUBE
> is using a different address to respond that it's not happy with.
>
>
> -nick
>
> On Fri, May 28, 2010 at 9:46 AM, Ryan Ratliff <rratliff at cisco.com>  
> wrote:
>> Not necessarily, but it will introduce a bit of complexity as the  
>> router has
>> to do the h.323 to sip internetworking.
>> In this case you should see CUCM send an H.225 Connect message to the
>> router, which should generate a SIP 200 OK out to the provider, but  
>> possibly
>> not until the router has completed H.245 media negotiations with  
>> CUCM.
>> If what I'm saying doesn't make sense it's time to engage TAC.
>> -Ryan
>> On May 28, 2010, at 9:38 AM, O'Brien, Neil wrote:
>>
>> Thanks Ryan,
>>
>> But the CUCM isn’t talking SIP at all though, the gateway is still 
>>  a H323
>> gateway.  Is this my problem??
>>
>> Thanks,
>> Neil
>>
>>
>> From: Ryan Ratliff [mailto:rratliff at cisco.com]
>> Sent: 28 May 2010 14:21
>> To: O'Brien, Neil
>> Cc: Frank Arrasmith; cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>>
>> It may be mostly semantics but at the point your router is doing IP  
>> to IP
>> it's acting as a CUBE.  Traditional voice gateways are IP to POTS.
>>
>> I'd disagree that your signaling is good however.  If the phone  
>> answers, and
>> that is not communicated back out the SIP trunk then something is  
>> not right
>> in the signaling.
>>
>> You need to look at the SIP messaging on the CUCM and at the  
>> router, just to
>> make sure everything CUCM sends is passed on.
>>
>> -Ryan
>>
>> On May 27, 2010, at 5:01 PM, O'Brien, Neil wrote:
>>
>> Hi Frank – it’s not a CUBE router.  Forgive my ignorance but  
>> should it be?
>>
>> Thanks,
>>
>> Neil
>>
>> From: Frank Arrasmith [mailto:frank.arrasmith at gmail.com]
>> Sent: 27 May 2010 18:40
>> To: O'Brien, Neil
>> Subject: Re: [cisco-voip] SIP Trunk on CallManager Gateway Router
>>
>>
>> This is just a shot, but it sounds like the signaling is good, so  
>> maybe
>> something with the media? Is your CUBE set for flow-through, or
>> flow-around?  When I set mine up, I had 2 problems, I messed with  
>> the media
>> settings, and changed to flow-around, which my equipment didn't  
>> support. It
>> should be flow-through, which I think is default.  The other  
>> problem I had
>> was with binding the media to an interface.  I either had problems  
>> with
>> routing to the interface, or bound the media to the wrong  
>> interface, I can't
>> remember off the top of my head.  Just a couple more areas to look
>> at...Please post back if you find a solution, as SIP/CUBE issues  
>> seem to pop
>> up more and more these days.
>>
>> --Frank
>>
>> On Thu, May 27, 2010 at 4:15 AM, O'Brien, Neil  
>> <nobrien at datapac.com> wrote:
>> 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 d 
>> on’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 t 
>> his...
>>
>> Thanks for reading!!
>>
>> Neil
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip