Hi, I think if your phone extensions are using SIP (signalling
protocol), then, they'll only be using the hosted PBX/Asterisk/SER to
find each other, and once that's done, the actual RTP packets travel
peer-to-peer. In a NAT environment (say your LAN is on a private IP),
i'm not sure if SIP can work effectively as it depends on how the the
phones registered it's contact point (IP address -- either the private
IP or the masq live IP), and that's another story.
Thanks!
Mark
Rehan Ahmed AllahWala - Super Technologies I wrote:
Hello,
We need some one who can tell us how to do this.
In a centrex / hosted pbx enviorement when a call is made from an extension to
extension
The call should remain within network, and the RTP packets should remain within
the network, once the ip is found.
So the internet is not actually used for an extension to extension call.
Please email me directly if u have a solution for this and charges.
Rehan
Super Technologies Inc., Pensacola, Florida
http://www.supertec.com - Technologies from tomorrow, TODAY!
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PSTN, SIP or IAX2 number, or Asterisk Superb Web Controls.
http://www.PhoneOpia.com - SIP Based OPEN Phone Services.
http://www.MySuperPhone.com - The NEXT Generation of Telephone.
Http://www.ip-pabx.com - Ip Centrex System, with global service.
Http://www.superPBX.net - One World, One Number, One Pabx, Physical.
http://www.didX.org - World's First DID Number Exchange / Peering Service.
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