A great book for you is the Cisco Press - Cisco Voice Gateways and Gatekeepers. I highly recommend it. Basically in the gateway you have to have both an incoming dial-peer and an outgoing dial-peer that match the Protocol, codec, dtmf, patterns, etc that you are using. The incoming dial-peer accepts the call into the gateway, the outbound Dial-peer directs the call to a destination. There is a lot to cover here, but incoming and outgoing do not have to be the same protocol, dtmf, etc, and this is known as a SBC if its IP to IP. If its Pots to IP, its just a simple voice gateway. The one thing to remember (and what I should have started with) is that there is a hidden dial-peer 0 on every voice router that can serve as your incoming dial-peer sometimes, but in the case of T1s/PRI's it won't answer correctly, so its best to create your own and set it up to answer all incoming calls ( . means all matches). Good luck with it and welcome to IP telephony.
Exactly what I was looking for. Thanks for the info. All the telephony books I have picked up revolve around call manager. unfortunately we went with a competitor, so not much of it applies. Ill be picking that book up shortly.
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The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
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Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...