Hello, new to asterisk and trying to set it up to work with my voip provider
(vbuzzer.com). I am behind a firewall that I don't have access to, to open
ports etc. Before using asterisk, I tried vbuzzer's windows client, and
linphone and twinklephone which all worked without having to enable nat or
stun. However I did have to enter the outboundproxy server to get them to
function. Not sure if it's an issue but my voip provider uses port 80 for
sip instead of 5060.
In asterisk, I can make outgoing calls through my voip provider to pstn
lines, audio works both ways. But calling in from my land line to the
asterisk box via vbuzzer, I get no audio either way. The local sip client
rings and when I answer the call, I see asterisk sending/receiving rtp
packets. I couldn't find much information on asterisk's outboundproxy
and
outboundproxyport variables. They were in chan_sip2 last year and then
merged with chan_sip. At that time, there were a glofal var, but now I
think they can be a peer. I then tried just having the asterisk server
answer incoming calls from vbuzzer, and I see it saying it's playing
monkeys, but no audio. Again, if I dial from within to the asterisk box via
a local sip client I get audio. I might be on the wrong track with the
outboundproxy, but since I'm not setting nat or stun in the other sip
clients and they can make and receive calls, I can't see what else it could
be.
I also read about siproxd, and it said in it's docs that (at that time)
asterisk didn't support outboundproxy and siproxd could be used as a
transparent proxy. Could siproxd be used behind the firewall as I am? I'm
running asterisk on my local box not on the firewall I'm behind. If someone
has experience with siproxd, I'd like to give it a try, but I don't see
how
to tie it in with asterisk and a voip provider.
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