[Asterisk-Users] sip phone extensions at a remote site

[Cameron Beattie]

If you're using SIP I think what you want is canreinvite=yes which means the 
two remote user clients can talk directly to each other. Asterisk disappears 
from the loop which means no accounting. I think NAT causes problems in this 
scenario also.

More details on the wiki

Regards

Cameron
----- Original Message ----- 
From: "cmould" <carey.mould at e2team.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, April 09, 2005 7:48 PM
Subject: [Asterisk-Users] sip phone extensions at a remote site


>I am in the proscess of integrating a clients remote and head office
> phone systems. Currenty each office has their own PBX and trunk lines. I
> am recommending that they put in an Asterisk server at the Head office
> with a WAN link to the remote office and switch to IP phones.  Trunk
> lines at the remote site would  be returned to the TELCO. External calls
> over the PSTN from the remote office would be routed over the WAN to the
> head office and through Asterisk to the PSTN trunk lines. All phones
> would then become extensions (both remote and head office locations). I
> want Person A in the remote office to dial an extension number and get
> Person B in the head office. What I am unsure about is if person A and
> Person B are both at the remote site and Asterisk PBX is at the head
> office, can A and B talk directly to each pther without traversing the
> WAN link? Has anyone done this before? What is the quality of the call
> if they have? Any information is useful.
>
>


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