Help understanding H.323 gateways (router) for PSTN access

[voip7372]

Please be patient with me, folks...as I come from an Avaya background and I'm now learning CUCM and how to setup a router (existing, with PVDM3, etc installed) to be a H.323 gateway for PSTN access.

I already have some working examples of the router config from colleagues but these are from places like Europe and other places where E1 circuits are used and it's not always easy to talk to my colleagues due to time zone differences, etc...so I'm hoping to learn some of this on my own, as much as I can...and combine it with what they're sharing.  Also, those H.323 gateways (routers) they're giving me info on are registered with an older CUCM (most of them anyway) and they're not using +E.164 for directory numbers like I will be using.  

Here's what I'm looking for.  What I would LOVE to have is a good example or configuration guide (with descriptions about the purpose for each configuration entry) of a router that's configured as a H.323 gateway for CUCM 8.6 or higher (if it matters...I'm on 10.5).  I'm looking for sort of a 'How To' type of guide for someone that is not deeply familiar with configuring the routers to work as H.323 gateways.   I've seen some Cisco documents, but they seem to go into every scenario under the sun and all I'm really interested in is the items listed below...

The basics of my config (so you know what I'm looking for):

• Router (3900 series) is located and used in USA
• Router will be H.323 gateway on CUCM 10.5 and used for PSTN access
• Router will have a 4 port ISDN PRI card (we'll be using all 4 ISDN PRI circuits...this is not E1, but T1)
• Directory numbers on CUCM are +.E164 format
• SRST is needed
• Router will of course need to transcode between the various codecs that might come into play as traffic gets passed over the WAN (like converting G.711 to G.729 and vice versa, etc)  It will need to be setup so that we can assign resources to the media resource group that will be used for the device pool associated with this site/router.

If you could point me to some info like I mentioned above, as soon as possible, I'd greatly appreciate it.  Thanks!

[Vivek Batra]

Friend,

I don't think you will get any single documentation which will help you to configure all what you have mentioned however I can try to redirect you to some useful posts which is atleast you need to know as per your requirements. Please note that this may not be enough to configure each and everything for your installation. You need to look at individual requirement and see what's best for you.

Configuring H323 gateway in CUCM and IOS;

https://supportforums.cisco.com/document/12231781/how-configure-h323-pri-and-gateway-cucm-10x

Voice translation in IOS

https://supportforums.cisco.com/document/48681/voice-translation-rules-media-gateways

Dial peers in IOS

https://supportforums.cisco.com/document/116861/dial-peers

SRST

https://supportforums.cisco.com/document/12212831/configuring-srst-cucm-10x

If you are using E.164 DN format in CUCM, only using translations in IOS may not sufficient since H.323 trunk doesn't support + and hence you may also have to look at calling/called number transformation under CUCM;

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/bccm-712-cm/b03trpat.html

Regarding transcoding, you didn't mention how many sites you have connected over WAN. If you have only one site, there is very rare chance that you would need transcoders. However FYR, you need to configure DSP farms accordingly in IOS to use transcoding resources;

https://supportforums.cisco.com/document/105611/how-configure-conferencing-and-transcoding-voice-gateways

Thanks

Vivek

[voip7372]

Thank you.  I will look at each of the links you shared.

Regarding E.164 and the +...so, you're saying that I might (for example) need to insert the + on the CUCM side?  So, incoming call enters the router via the ISDN PRI circuit and then we'll need to insert the + once the call hits CUCM?  Assuming that the only calls entering CUCM from the router are coming from PSTN, we could insert + on the gateway config in CUCM?  (every calls gets + inserted as the calls enter from the H.323 gtwy).

Just curious.  Is there any reason why we would want to make this a MGCP gateway rather than H.323?   Given our requirements for SRST, etc...is there a compelling reason to make the router a MGCP gateway vs. H.323?  Or would it be more appropriate and flexible to make it a H.323 gateway?

EDIT:   Or...would it be wise to configure the router as a SIP gateway/trunk?  I didn't even think of that until just now when I did a search and found someone mentioning the config to setup a router as a SIP trunk.  Would the config be fairly similar to the H.323 config?  Sorry for my basic questions, but like I said, this is mostly all new to me.  I'm pretty good on the server side now, but the router side is what I need the most help with.

[Vivek Batra]

Regarding E.164 and the +...so, you're saying that I might (for example) need to insert the + on the CUCM side?  So, incoming call enters the router via the ISDN PRI circuit and then we'll need to insert the + once the call hits CUCM?  Assuming that the only calls entering CUCM from the router are coming from PSTN, we could insert + on the gateway config in CUCM?  (every calls gets + inserted as the calls enter from the H.323 gtwy).

I assume that when you are saying your DN will be in E.164 format, so it would be something like this +1-408-XXXX-XXX. If yes, how are you planning for users to make internal calls as it won't be feasible for internal users to dial E.164 number to reach internal destination. You would need some sort of translations inside CUCM.

Anyway... while receiving call from gateway to CUCM (incoming call on T1) and assuming you don't put any number translation in gateway, whatever called number you get on H.323 trunk somehow to be manipulated to E.164 for call to successfully see the desired DN.

Depends on the called party number format you get during incoming call, it may not be feasible to add + straightaway for all incoming calls. Your provider may be giving 7 digits for local calls, 10 digit for long distance etc. Hence you need to add prefix (+ or area code or national code) accordingly.

Just curious.  Is there any reason why we would want to make this a MGCP gateway rather than H.323?   Given our requirements for SRST, etc...is there a compelling reason to make the router a MGCP gateway vs. H.323?  Or would it be more appropriate and flexible to make it a H.323 gateway?

There is no doubt that MGCP is simpler to configure as you don't need to play much with gateway however when it comes to features, H.323 is preferable specifically with SRST. For instance, if you use MGCP, you don't need to configure dial-peers (MGCP doesn't use dial peer) and translation rules etc in gateway however just because to take care of SRST, you would have additional configuration in gateway for MGCP to fallback to H.323 during SRST. In result, you end up with configuring dial-peers and translation profiles/rules in gateway. Anyway you are configuring dial-peers and translations in gateway even with MGCP, why not to use H.323 then which is more scalable and use the full advantage of dial-peer and translation rules from the beginning.

If SRST is not required and you are good to go with no calls during WAN failure (obviously you would want to call still go during WAN failure), MGCP may be a good choice.   

Thanks

Vivek

[voip7372]

Correct.  Our internal DNs are in the format +17145551212, for example.  I believe the Telco is sending only 4 digits.  We'll assume that in CUCM (on the H323 gateway config page for this router) for any incoming call to CUCM from this H323 gateway, we'll need to insert +1714555 to complete the actual DN that needs to be called.  

For people to call internally to each other, we'll have local translation patterns in a PT that's assigned to that site's CSS (for those phone devices there) so they'll be able to 'short dial' to the ext, like 1212 (translation pattern will insert +1714555).  

I can't think of any type of INCOMING call we'd be getting from this H323 gateway other than calls from the Telco that are intended for an internal +E164  number in CUCM...so I assume it would be safe to insert the +1714555 (in this example) right there on the H323 gateway page in CUCM.

Ok, we do need SRST, so it sounds like H323 is the way to go.

[Vivek Batra]

I can't think of any type of INCOMING call we'd be getting from this H323 gateway other than calls from the Telco that are intended for an internal +E164  number in CUCM...so I assume it would be safe to insert the +1714555 (in this example) right there on the H323 gateway page in CUCM.

If you are sure irrespective of the call type viz local, national etc, your provider will always send you last four digits only, what you are saying should work.

For people to call internally to each other, we'll have local translation patterns in a PT that's assigned to that site's CSS (for those phone devices there) so they'll be able to 'short dial' to the ext, like 1212 (translation pattern will insert +1714555).  

Perfect.

Ok, we do need SRST, so it sounds like H323 is the way to go.

SIP is another choice for you.

Thanks

Vivek

[voip7372]

If you are sure irrespective of the call type viz local, national etc, your provider will always send you last four digits only, what you are saying should work.

Yes, it's just ISDN PRI and the phone company will always be sending the last 4 digits if the phone number.  I was hoping it was as simple as I thought it would be...just insert the +1714555 (or whatever the case may be) right there in CUCM on the gateway page - "Call Routing Information - Inbound Calls" section of that page in the Prefix DN field.

SIP is another choice for you.

I guess we should stick with H323 since that's what my other colleagues are already familiar with and I'd be able to get more internal support from them if we stick with H323.

[Vivek Batra]

Yes, it would be simple. Just make sure that you put +1714555 as prefix in all number types as you may not know what number/plan your service provider may be including in SETUP message. Alternatively, you can use show isdn status command in gateway to get the desired plan/type info. 

Thanks

Vivek

[voip7372]

Oh, wait.  So, you're saying to use the "Incoming Calling Party Settings" section, not the "Call Routing Information - Inbound Calls" section, correct?  See screenshot for my example and notes.  I assumed I'd be using the Prefix DN field:

Prefix DN

Enter the prefix digits that are appended to the called party number on incoming calls. Cisco Unified Communications Manager adds prefix digits after first truncating the number in accordance with the Significant Digits setting. You can enter the international escape character +.

Sorry. I meant to fill out the CALLED party section in my attached screenshot/example, not the calling party.

[Vivek Batra]

Yes, I was referring CALLED party section.

I have never used Prefix DN although as per feature definition, it should also serve your requirement.

Thanks

Vivek

[voip7372]

Ok, thank you.

[Vivek Batra]

You're most welcome.

Thanks

Vivek