As a prelude to a full VOIP implementation with CallManager, we'd like to be able to do some testing by interposing 2 2600 routers with 2*PRI each between the PSTN and our PBX.
The notion is that we'd ship the voice traffic from the PSTN across a FE interface to the other router which would spit it out on the PRI's towards the PBX and vice versa using dial-peers.
PSTN <2*PRI> 2600 <FE> 2600 <2*PRI> PBX
The notion is that we'd thus be able to implement additional VOIP devices in the network and test using dial-peers while not impacting the existing function of the PBX. We'll use a x-over FE between routers for the 'normal' operation via the PBX, and the other FE's out of the 2600's for VOIP testing with other devices.
Does this sound like a plausible strategy ? If so, could anybody provide a generic dial-peer config that would basically accomplish this ?
You could do that, but I would suggest buying a single copy of Call Manager, a server, and a couple IP phones to go with the 2600s. If you are planning unified messaging you can also get a free 2 port demo version of Unity that will run on almost any PC.
Your proposed setup will teach you about H323, and QOS, which will but useful, but there is much more to learn if you are heading for a full IP telephony install. We did a pilot project like this, and it was invaluable in preparing for the full deployment.
Here is a good doc on dial peers and VOIP transport:
Yes, we are planning a full VOIP implementation with Call Manager and Unity.
The initial notion however is that this system would allow us to phase in equipment that would let us do additional VOIP testing while limiting the impact on the existing voice traffic by keeping the amount of potential failure points as low as possible.
I understand where you are comming from with a phased approach with your install. My only question is what are you planning on accomplishing once the two routers are online (that it works)? I would also recommend a small Call Manager and some inexpensive IP phones so that you can demonstrate the system.
IntroductionCUCM Routing RulesDial String implementation PolicyCUCM Routing LogicSIP URI Call Routing Analysis+++ Case Study: 1 ++++++ Case Study: 2 +++Conclusion
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