standalone vc cant call sip address?

[baselzind]

i have a stand alone vc that have a private ip that is natted into a public ip for calling and receiving calls , i have sip disabled for a certain issue only h.323 enabled for calling , i have this sip address that i cant call , could it be that i have sip disabled that i cant call them ??? whenever i call the address i get this message "your system isnt registered to the gatekeeper" ??? by the way i can receive calls from them succesfully?

[Joe Vallender]

Is your VC unit registered to an H.323 gatekeeper?  If so, what do the gatekeeper logs say when you dial the failing address?  Does your gatekeeper know how to route to the failing address?  Could be, the gatekeeper needs to have the failing addresses' own gatekeeper/SIP server/VCS IP address set as a gateway/route to the address.

If your unit is not registered to a gatekeeper, then how does it know a route to the failing SIP address?  If its call is actually reaching the far end SIP server via DNS lookup, then possibly that server/gatekeeper/firewall is blocking the completion of the call because the originating address, of an endpoint not registered locally, is not allowed to complete calls into the servers network.  Outgoing calls may not be blocked as seen by your VC unit able to receive calls from the SIP endpoint.

[baselzind]

there is no gatekeeper , my vc unit just connect to the firewall and use natting for a public ip address , i tried calling a loopback address and it worked so i guess my vc unit is okey.

[Jens Didriksen]

You need to provide more information: type of end-point you are using and software version? SIP address format you are trying to call?

Have you tried enabling SIP to see if you can connect to that address using SIP?

I assume this other party is calling your system using the public IP address of your system, in which case it should be showing up as a H.323 call.

Calling a SIP only address using H.323 will not work, as these are two different protocols, unless there is a CAC device such as a VCS in the mix which can interwork (translate) the call for you.

/jens

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[baselzind]

the device im using is C40 , version TC5.1.4.295090 , their address is like this XXXX@XXXX.com

[Jens Didriksen]

That software version is seriously old considering C-series is now up TC7.3.6 - won't help with this issue though.

So, since their SIP URI is alias@domain, their system must be registered with a CAC device such as a VCS, CUCM etc.

They will also need to have SIP SRV records in place pointing to the device handling the calls, otherwise any SIP calls to alias@domain will not work.

Now, they also need to have H.323 SRV records in place, if they want to allow H.323 calls at all, as incoming H.323 calls will otherwise not know which device is handling the call, and the call will not succeed, Remember, SIP and H.323 use different ports for call setup etc.

Having said that, depending on their deployment, they might still need to translate the incoming H.323 call to SIP before it is sent on to their end-point(s).

As I said before, at least enable SIP for a short while so you can make a test call using SIP, if that connects, then you will at least have narrowed the issue down considerably.

/jens

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[baselzind]

you mean this particular address  need to have a SRV record? and a H.323 SRV record to direct incoming h.323 calls? btw the adddress looks more like this XXXX@XXXX.domain.com

[Jens Didriksen]

Doesn't matter if it's xxxx@xxxx.domain.com or xxxx@domain.com - any URI type address using that kind of format must have SRV records in place pointing to the device handling the calls.

When your system attempts this kind of call, it will do a DNS lookup asking "who looks after domain.com?" and the query will be forwarded by high level DNS servers to the relevant DNS server for that domain and its sub-domains.

This DNS server must have the relevant record in place in order to send the call to whichever device is handling the call. The call will go nowhere if these records are not in place, and will normally return "404 not found" type error,

SIP uses ports 5060 and 5061 for call set-up purposes, and the SIP SRV records will reflect that.

Call set-up ports using H.323 are 1719 and 1720, so you see how this call will not go anywhere if you are calling a "SIP only" destination, or if they do accept incoming H.323 Annex O type calls (URI), if they don't have H.323 SRV records in place.

To bypass any DNS issues, you would have to call alias@Public_IP_address_of_CAC_device (CAC = Call-Admission-Control)

/jens

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[Joe Vallender]

If you provide the domain of the failing VC unit's SIP address, anyone with a DNS Lookup tool (VCS Admin) can request a public DNS evaluation to see if there are H.323 SRV records and what they point to.  This will confirm if your H.323 call can even start to get to a destination.  You can also try calling Verizon's video conferencing access test 'connectcheck@join.verizon.com' which accepts public SIP and H.323 calls from the internet. Depending on the originating device's H.323 dialing capabilities you might have to use '63.97.112.115##1866845273343214321' or appropriate vendor's direct IP dialing format instead of the URL.

[baselzind]

is the dns lookup tool on the vcs-e admin page or it a seperate website? also as the vcs address is like this xxxx@xxxx.domain.com do i check xxxx.domain.com or domain.com only?

[Joe Vallender]

I don't currently have access to a VCS, but from memory, it is in the Maintenance tab.  From a VCS 8.7 Admin Guide it is in Maintenance->Tools->Network Utilities->DNS Lookup

[Patrick Sparkman]

To check if you have the required DNS records for URI dialing, you can use the VCS to perform a DNS lookup as mentioned by Joe, located under Maintenance > Tools > Network utilities > DNS lookup.

As an alternative, you can use Cisco's VCS Service Checker.  It's meant for those that want to check inbound calling to their VCS/Expressway, but standalone endpoints that want to use URI dialing in the format of alias@domain will need to have the same basic H323/SIP records as the VCS/Expressway.

To understand more about SRV records and how they play a part in URI dialing as well as what to look for, refer to the DNS SRV for TelePresence Video Communication Server.  If you plan on using alias@domain, you'll need SRV records, but without a gatekeeper to register your endpoints to, you can only have one endpoint tied to a single SRV record, if you did have a gatekeeper such as a VCS, you could use the same domain for all endpoints and just have different aliases for each.

[baselzind]

but do i check using the dns tool xxxx.xxxx.com or xxxx.com only , for the vc xxxx@xxxx.xxxx.com address

[Patrick Sparkman]

You'd check the entire domain as it appears after the @ sign, so in your example it would be: xxxx.xxxx.com.

[baselzind]

i checked it and there was zero records? does that mean the vc  cant receive sip calls in the for of alias@domain.com? only through public ip address?