Asterisk Gurus

[chpalmer]

I havent done an exhaustive search on this but havent found the post I am looking for...

It seems when I play with my Asterisk based box here that there are two different servers involved on Voipo's end.

And- if I set it up one way I get one incoming server in my firewall states. But another if I set it up "Nat + Dynamic IP... Picture at "Topology"

I think my outgoing calls dont work using their product because this particular product does not have settings to deal with this architecture...
Am I on the right track here...

[dswartz]

I'm not sure I'm reading your post correctly, but it's hard to tell exactly what you mean from the limited context you posted. My config (asterisk 1.4 + freepbx) has only one voipo server for the trunk - sip.voipwelcome.com. Why do you have two?

[chpalmer]

When I use an ATA, I see only the one server in my firewall states.

When I try to use an Asterisk product it connects to one at the sip port (say 5060) but when a call is made I connect out to that same server but another tries to connect back this way...

My firewall states show this way...

65.xx.xx.xx:52854 <- 172.31.125.21:49518

172.31.125.21:49518 -> mypublicip:56758 -> 74.52.58.50:52854

I think this is by design..?.

[dswartz]

When I use an ATA, I see only the one server in my firewall states.

When I try to use an Asterisk product it connects to one at the sip port (say 5060) but when a call is made I connect out to that same server but another tries to connect back this way...

My firewall states show this way...

65.xx.xx.xx:52854 <- 172.31.125.21:49518

172.31.125.21:49518 -> mypublicip:56758 -> 74.52.58.50:52854

I think this is by design..?.

Ah, okay! I get it. Yes, this is by design. I'm about 99% sure what you are seeing is the switch in question connecting back to you. Unlike some providers (viatalk?) who funnel everything through the server (SIP and audio streams), voipo does not. Whichever switch handles your call is what connects to you to process the audio stream...

[chpalmer]

Cool- Thats what I thought!


Thank You!!

[VOIPoNorm]

SIP Signaling is usually on port 5060.

RTP (the actual audio stream) is usually on the higher numbered ports.

There are lots of extra details, but the above is from the helicopter view.

Regards,
Norm