[cisco-voip] Use Gatekeeper for Tall Bypass

[Ryan Ratliff]

If your phones are in SRST then they have lost IP connectivity to the CUCM servers.  Most people assume that whatever caused this is also going to cause the branch to lose IP connectivity to the other branches as well, thus dialing through the PSTN is the only way to actually get calls out.

You can certainly come up with a complicated dial plan for SRST so it can try routing calls to other branches via voip and then fall back to the PSTN but before putting that effort in you really need to consider if it will ever get used.

-Ryan

On Feb 21, 2010, at 5:47 PM, Manoj Kalpage wrote:

We use "0" for PSTN dialing. Can I still use translation patterns when use "0" for outgoing call? I thought translation pattern is for incoming calls.
 
This is what I want to do.
 
Branch A Phone B extension : 75341212
Branch A Phone B DID : 0345671212
 
Branch C Phone D extension : 75351313
Branch C Phone D DID : 0457731313
 
When Branch A Phone B call to Branch C phone D via PSTN, phone B dial 00457731313.
I need these calls convert to extension number (75351313) and route through VoIP.

Thanks,
MK
On Mon, Feb 22, 2010 at 4:07 AM, Ahmed Elnagar <ahmed_elnagar at rayacorp.com> wrote:
I think yes it is better to use CM translation patterns for this, you can have one translation for each site that match DID range for remote sites and convert it to internal DN.

 
  Best Regards;

  Ahmed Elnagar

  Senior Network PS Engineer

 
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  Phone: +202 3827 6000 Ext.2475

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  E-mail: ahmed_elnagar at rayacorp.com

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From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Patrick Fischer
Sent: Sunday, February 21, 2010 7:25 PM
To: Kalpage Manoj Perera
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Use Gatekeeper for Tall Bypass

 
Hi MK

 
There are several ways to solve this issue. Just think about translation patterns. But yes: a GK could also do the trick.

 
Regards

Patrick

2010/2/20 Kalpage Manoj Perera <manoj.kalpage at gmail.com>

Hi All,

We have central CUCM cluster and 50 SRST sites. We have announced users to use extension dialling when call between SRST sites and head office but they still call via PSTN. I am wonder we can deploy gatekeeper and automatically re-route PSTN call for SRST branch offices to VOIP. Is this possible with gatekeeper or any other piece of hardware ?
Any information would be greatley appreciated.

Thanks
MK

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