[plug] Sip to Sip

[William Kenworthy]

moriah PXE # cat /etc/asterisk/rtp.conf 
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=7070
rtpend=7099
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000   ; Milliseconds between rtcp reports 
                        ;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that
; do not come from the source of the RTP stream. This option is
; disabled by default.
; strictrtp=yes
moriah PXE #

for a small system, 10 ports is plenty

If using a linux firewall, look into the SIP netfilter module

BillK



On Wed, 2010-10-27 at 18:44 +1000, Tim 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
> >
> 
> 
> 

-- 
William Kenworthy <billk at iinet.net.au>
Home in Perth!