[Asterisk-biz] Consutant Needed: Direct ip 2 ip call in a centrex

[Mark John Buenconsejo]

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!
>http://www.VirtualPhoneLine.com - Get A US, UK, EU Number, Forward it to 
>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.
>
>
>_______________________________________________
>Asterisk-Biz mailing list
>Asterisk-Biz at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>  
>