[plug] Sip to Sip

Tim weirdit at gmail.com
Wed Oct 27 16:44:31 WST 2010

Now I feel silly.
I forgot to open firewall ports for the RTP streams.
The asterisk site says to open ports 10000 to 20000. Now I know I
don't need that many RTP ports. Anyone have an idea where it defines
the ports so I can limit it down a bit?



On 27 October 2010 18:25, Tim <weirdit at gmail.com> wrote:
> 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

Timothy White - Somewhere in Australia

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