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Need the experts advice on IPT design and installation.

Hi ,

I need some advice on the IP telephony design I have to install shortly. Following are the details of the hardware at my HO site

1. Existing Alcatel PBX that serves some 400 analog / digital phones

2. Existing E1 ISDN PRI with 30 channels + 400 DID numbers configured on the Alcatel system.

3. Existing 2 meg WAN lease line to a remote branch office.

4. A new callmanager cluster ( 2 servers ) that will provide IP telephony solution to the new users ( around 300 IP Phones ) at the same HO location.

5. A new voice PSTN / PBX gateway 2801 that will have 2 E1 MFT ports that will be used both PBX and PSTN integration on E1 ports.

6. A new Unity server for only providing voicemail to IP Phones users at HO and also the remote branch office.

7. A new E1 ISDN PRI with 30 channels + 600 DID numbers that will be configured on the callmanager for the new IP Phone users ( 300 users at the HO and some 100 IP Phones for remote branch )

Following are the details of the hardware at my remote branch office.

1. A voice gateway 3845 with SRST license for 720 users with an AIM-CUE module to provide voice mail for 50 users.

My question is

1. Can I use the same DID number range for the remote branch IP phones. i.e Can I configure the first 300 DID block for IP phones at the HO and the next 300 DID numbers for the IP Phones at the remote branch.

2. Will there be any changes in the configuration on the CM for the DID block for remote branch across the WAN.I want to have toll free calls to my remote branch on the same WAN link.

3. Is there a solution by which I can configure the AIM-CUE module for voicemail serives to the remote branch IP phone users only during WAN link failure. i.e CM and the Unity server are down. At all other times, the users at the remote branch should get their voicemail form the Unity server at the HO.

4. Can I assume that the per call bandwidth across the WAN can be around 12Kbps + headers .

5. What will be the bandwidth used for a typical voicemail acoss the WAN.

6. Is there a configuration guide by which I can configure the router at the remote branch to work as SRST during WAN link failure to provide PSTN and Voicemail services to the users.

7. Whats the recommended configuration to create a trunk with the Alcatel PBX system to achieve transparent calling between the two systems.

Thanks

Brgds

Renzil

2 REPLIES
Silver

Re: Need the experts advice on IPT design and installation.

As far as SRST isd concerned, you can configure it at the remote end. Here's the URL:

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_administration_guide_chapter09186a00801f3ac8.html

New Member

Re: Need the experts advice on IPT design and installation.

Answers:

1. Yes you can have one DID range on the HO Gateway and another DID range on the branch office. Careful planning with Partitions and Calling Search Spaces will accomplish this. If you want the DID's to come in across two seperate T1's and Gateway's you will have to coordinate with your telco on this. Obviously, you can't split a DID range for a 404 area code and have it ring down some gateway in another area code without some special programming from your LD carrier.

If you only have 1 DID block in the HO and want the calls to ring the branch, then I would send the calls across the WAN via g729 or something. Be carefull with 911 and caller-id.... blah blah blah etc.

2. It's hard to understand what you want EXACTLY, but based on what it sounds like you will definately have the ability to configure the call manager to interoffice call across the WAN. Even making toll-bypass calls to other numbers in the remote office area code is possible with a local gateway.

3. I don't know about using an AIM-CUE module, but I know that Cisco has an internal document on how to do Unity-SRST when a WAN link fails. It basically requires an independant router running CME that registers backup ports to the Unity Server.

4. Depending on the codec. I would use Cisco's tool at: http://tools.cisco.com/Support/VBC/do/CodecCalc1.do

5. Unity can be setup to play g.711 or g.729 prompts. So again this will depend on the codec used.

6. There are certainly SRST guides for failover PSTN, but I know I got my UnitySRST document from Cisco who said it was an internal document. I can't speak to the AIM-CUE solution with CallManager. I am going to go out on a limb here and say it will not work.

7. Trunk to the old PBX will be pretty easy. You might want to make a seperate access code to use that trunk, but it really just depends on your dial plan and overlap.

Sounds like a fairly complex configuration. The good news is that it only has 2 sites. In depth knowledge of Codecs, DSP resources, Calling Search Spaces, Route Filters, and Partitions is going to be important to make this successful.

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