[plug] Sip to Sip

Scott Middleton scott at assuretek.com.au
Wed Oct 27 17:24:50 WST 2010


It has changed a bit over the years. On my elastix box it is.
/etc/asterisk/rtp.conf

some notes:
rtp is udp so its no big deal having that many open ports
sip 5060 is tcp and udp

Generally i port forward udp 10,000-20,000 and tcp/udp 5060 to my asterisk
server. But I only allow it from my VISP.

Scott Middleton
Managing Director
Linux Consultants Pty Ltd t/as AssureTek
Email - Scott at assuretek.com.au
Phone - 1300 551 696
Mobile - 0400 212 724



On 27 October 2010 16:44, Tim <weirdit at gmail.com> wrote:

> 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?
>
> Thanks
>
> Tim
>
> 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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>
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