Is it feasible to use multiple VG350's to support a couple of hundred analog phones?
This question is intended for those that have a good bit of experience with Cisco VoiceGateways and possibly old style PBX's. To give a summary of my dilema, I have a few hundred analog phones that I would like to integrate into my VoIP network to wean us off of our PBX. I'm looking at something around 200 phones that would require the swap and cannot be replaced with VoIP phones easily; i.e. they are outdoor, connected to elevators, or have no Cat5/6 infrastructure near them.
I'll add that I'm not terribly well experienced with telephony and VoIP, so this question may be off, but would it be feasible to run to VG350's to support these phones?
On top of that, do VG's share information with each when they have a direct connection, in the way that a router might broadcast its available routes to antoher router? Or would I have to provide a route for every single phone, or range of phones, connected to the first VG into the second VG and vice versa?
Would the VG's should the processing burden of translating between VoIP and analog, or would this create overhead for the CUCM?
You did not specify what your PBX is going to be, so I will assume it's CUCM, then to answer your second question CUCM has the dial-plan defined for your analog extensions as it's in charge of routing calls to/from it, the analog GW without a call agent such as CUCM cannot do anything.
Depending on the protocol you use (SIP, H323, MGCP, SCCP) the configuration of dial plan is different, if using MGCP/SCCP you defined the analog device similarly to IP phones in CUCM, if SIP/H323 then you define patterns to them and point to respective trunk and then configure the GW with dial-peers. Decision on protocol boils down to what type of analog devices and feature you have, i.e. for faxing to get full T.38 protocol-based v.3 support you'd need SIP/H323, for analog phones to get supplementary feature such as hold, etc only SCCP can do that.
With SCCP the routing would look identically to routing from IP Phone where the CSS/Device Pool (if using local route group) would specify the routing.
As to the RTP it would be between the analog GW to whatever the end device is, i.e. IP phone, or PSTN GW, so the traffic from analog GW to far side would be IP, and traffic from this analog GW to the analog device is the analog across the analog wire.
IntroductionCUCM Routing RulesDial String implementation PolicyCUCM Routing LogicSIP URI Call Routing Analysis+++ Case Study: 1 ++++++ Case Study: 2 +++Conclusion
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