[plug] Sip to Sip

[Tim]

Thanks to Chris I've been playing around with Asterisk.
I'm playing with call files, which make it easy to initiate a call
from command line.
Found this article after basically working out this method myself.
http://superuser.com/questions/123899/asterisk-trying-to-use-call-files-to-create-a-conference-call-between-two-dynam

How ever, while this works in creating 2 calls, no audio appears to go
between them. i.e. I pickup one phone, someone else picks up the
other, and we hear nothing between them. Hanging up the second phone
correctly terminates the first, just no audio between them.

Anyone experimented with this kind of thing, or have Asterisk
experience and have any ideas. Please let me know.

Tim

On 27 October 2010 14:14, Tim <weirdit at gmail.com> wrote:
> I've asked this years ago I think, but I'll ask again.
> I'm looking for a way to make 2 outgoing SIP(VoIP) calls that are connected.
> i.e.
>
> Server calls Client 1
> Server calls Client 2
> Client 1 and Client 2 are connected, via the server. There is no
> talking party on the server.
>
> Anyone know how to do this with Asterix or Freeswitch or another linux tool?
>
> Thanks
>
> Tim
>
> --
> Timothy White - Somewhere in Australia
>



-- 
Timothy White - Somewhere in Australia