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

[stotaro]

SIP=Session Initiation Protocol
canrinvite=yes


> 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
> >
> >
> >
>
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>
>
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