I hear often that it is hard to practice/demo voice related configurations with GNS3. This is only partially true.
It is correct that GNS3 cannot substitute Multiservice routers like 28xx etc because the DSP resources cannot be emulated. For using T1/E1 or FXS/FXO connections you need the real hardware interfaces and of course another PBX/router interface to connect your T1 to. If you have a double T1 you could use a T1 x-over to let the router do both ends.
For all other configurations like Dial-peers, translation patterns, SIP-UA configurations, CME, ephone registrations, CUBE etc, GNS3 can do the job. Just an example from my own lab: I have several accounts at SIP-Providers in US and Europe to use with a standard sip-phone like X-Lite. The router can log in this SIP-account and let CME or UCM handle the call. Via this way I provide a local german PSTN-number for my german friends which makes my phones in Minnesota ring. The thing has of course a downside… they keep forgetting that I am 7h behind, but for this purpose Unity will take the call.
Where is GNS3 now coming into the ballpark? For each router you can specify only one SIP-registrar server, but I like to have all my SIP-accounts registered at the same time to route calls between them. GNS3 can now run the virtual routers, one for each SIP-provider, register your PSTN-numbers and UCM/CME can do the call handling.
by Patrick Geschwindner, Ascolta











