So here’s the theory. You have a main office with a master asterisk server (Its on a Static IP). You want to add VoIP connection to a small remote office on DSL (Dynamic IP). I wanted to test this out for a while, I finally got the chance. In the extended contents I’ll run down the configuration (thats alot easier to understand than [this][1]) & an easy to test demo.
[
][2] To the left is the over simple picture of the world. This is the setup that I was using since at the moment there was phones attached to the slave server (remote office). I’m going to configure a block (in this case 500-519) of extensions to point at the slave server. On the slave server extension 500 is going to be an echo test & 510 will be a talking clock test (of course you could point these at phones — No problem).
Master — IAX.conf
<br />
[SlaveServ]<br />
type=friend<br />
secret=x123x<br />
context=inbound<br />
host=dynamic<br />
Master — Extensions.conf
<br />
exten => _50X,1,Dial(IAX2/MasterServ:1234pass@SlaveServ/${EXTEN},30,r)<br />
exten => _51X,1,Dial(IAX2/MasterServ:1234pass@SlaveServ/${EXTEN},30,r)<br />
Slave — IAX.conf
`
;Somewhere under [general]
register => SlaveServ:[email protected]
;32.12.31.52 – Being your Master Servers IP or HostName
;At the bottom of the file
[MasterServ]
type=friend
secret=1234pass
host=dynamic
`
**Slave — Extension.Conf**
`
exten => 500,1,Playback(demo-echotest)
exten => 500,2,Echo
exten => 500,3,Playback(demo-echodone)
exten => 500,4,Hangup
exten => 510,1,Answer
exten => 510,2,SayUnixTime()
exten => 510,3,Hangup
`
Now restart Asterisk on both servers (give them about 1 min to sync) — Pick up your phone attached to the Master Server, dial 500 — and you should get the echo test. In my case the Slave was behind a linksys — but because it registers to the Master — No port forwarding was needed.
[1]: http://www.voip-info.org/wiki-Asterisk+-+dual+servers
[2]: /uploads/ferrarisimpleaster.GIF