[cisco-voip] two clusters - same gateway sanity check

[Leslie Meade]

Yea I have done something like this before.
Used H323 gateways for termination, used an Intercluster Trunk to route calls into the new cluster, and then H323 back out to the pstn.
Cannot see any issues changing it up from H323 to SIP for the out bound calls.



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Wellnitz
Sent: Wednesday, May 16, 2012 1:31 PM
To: cisco-voip
Subject: [cisco-voip] two clusters - same gateway sanity check

Okay...I've been thinking (it happens once in a while) over some things and wanted to do a sanity check.


1. Instead of using H323 to a gaateway has anyone had success using sip?  Would this require the use of MTP?  How would this impact SRST?  I know MGCP would be ideal but then we lose our ability to monitor PRI traffic until I can justify dishing out the cash for Cisco's UC management suite.  Will SIP allow us to receive calling name from the PSTN in a  similar fashion as h323?  In my viery vivid imagination this should jsut be a simple SIP trunk from CM to the GW.

2. If I use h323 to/from the old CM cluster would I feasibly be able to use SIP from the new cluster simultaneously then move inbound over to SIP at a later date?

Initial Outbound Scenario:

CM Cluster 1 ----->h323---
                                     |
                                      --->39XX Gateway----->PRI
                                     |
CM Cluster 2 ------>SIP----

Initial Inbound Scenario:

CM Cluster 1 <-----h323<------39XX Gateway <----PRI
       |
       |
      V
CM Cluster 2

My main goal is to be able to completely test the new cluster without impacting end users at all.   I could do the same scenario with h323 but I'd rather use SIP because it would make things very easy as we migrate to SIP trunking for PSTN access.  All I have to do then is point the same SIP trunk to a CUBE instead of messing with call routing on the CM.

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