Hello, new to asterisk and trying to set it up to work with my voip provider (<a href="http://vbuzzer.com">vbuzzer.com</a>). 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.
<br><br>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.
<br><br>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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