I'd like to hear opinions on this scenario PBX-*-remoteIPpho


I need to connect an small office based in Geneva to their main office’s PBX in Panama. Professional communication is the main goal, I shall be generous on the design though.
They require at least one simultaneous call to be near pstn quality.

I am thinking in setting up an Asterisk for this, but would really like to know other opinions and experiences on similar scenarios.

First I’ll connect 4 analog extensions of the PBX to an analog board probably Digium, but has anyone done it with a Dialogic board? (I have some spare)

then all calls that come on those extensions will be redirected to remote IP Phones, so that they are like a “part of the local PBX”, and can be reached individually from anyone on the PBX.

I was thinking on creating a VPN (using OpenVPN) in the server in Panama (which has public IP of course) and a remote linux machine in Geneva (with a dynamic IP) which connects to the VPN to extend the network to their office there (so to have access to printers file servers,etc) and also to be able to use SIP phones without too much firewall hassle.

I can also set up an asterisk in Geneva and connect them both via IAX2, but don’t know if this would improve my scenario performance in anyway (other than possibly connecting to Geneva’s PSTN)

Internet connections in both ends can be increased, it is currently in 512kbps. Only one simultaneous call is needed average (two at most but very seldom).

I was thinking in using GSM as the codec, but could consider buying G729 licenses.
With phones we are considering Polycom or Snom

Does anyone thinks this would work, is there going to be too much echo or lattency or jitter to be able to work properly?
(there is at least one jump to the sattelite) considering the first leg of the comunication from Panama to US goes through fiberoptics.

If so, what could be an advice to minimize these possibilities.

I welcome any kind of comments or advices, I haven’t started yet

Also let me know if I’m not in the right ballpark to even try this.

any coments anyone?

1 call needs maximum 64 kilobits - especially if you will use ulaw alaw codecs.
More important is whether quality of Internet connection is good?
It is better to use GSM or ulaw, alaw codecs - less troubles.
If there are some echo - you have a lot of opportunities to suppres it - depends manly from computer.

thanks for your comment,

How about, the choices of having one Asterisk and the remote end with just SIP phones, against having a remote Asterisk and interconnect them via IAX, is there any performance issue (for better or worst?)