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How can one test VoIP functionality with their existing PBX or Key system?

How can one test VoIP functionality with their existing PBX or Key system?
There are multiple possibilities for testing VoIP functionality with an existing PBX or Key system. How you test depends upon your goal.

If you have two sites linked together with PBX tie lines and you want to try using VoIP so that calls will be routed over your internal network rather than costly tie lines, you can test using a SIP to PSTN gateway (such as the MX25.)

This configuration could look like this:

Existing PBX <- T1 PRI -> MX25 <- SIP over WAN network -> MX25 <- T1 PRI -> Existing PBX

Perhaps you have a single site and you want to keep your existing PBX and connect long distance calls through an Internet telephony service provider that provides superior rates. In this case, you could use a SIP to PSTN gateway and connect in this fashion:

Existing PBX <- T1 PRI -> MX25 <- SIP over Internet -> ITSP ->

Perhaps you are planning on replacing your legacy PBX and putting in an IP PBX (such as the MX250) to test the functionality before cutting over service. In this case, the configuration could look like this:

Existing PBX <- T1 PRI -> MX250 <- T1 PRI -> PSTN

Using this approach, the existing PBX continues to function as it always has and only dial plan entries are required to route calls between systems. This allows for certain employees to learn the new VoIP system and understand its features before migrating over service.

This was last published in November 2004

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