[plug] Sip to Sip
billk at iinet.net.au
Wed Oct 27 17:27:15 WST 2010
moriah PXE # cat /etc/asterisk/rtp.conf
; RTP Configuration
; RTP start and RTP end configure start and end addresses
; Defaults are rtpstart=5000 and rtpend=31000
; Whether to enable or disable UDP checksums on RTP traffic
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
; 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.
moriah PXE #
for a small system, 10 ports is plenty
If using a linux firewall, look into the SIP netfilter module
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?
> 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!
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