Need help to end my setup.. i´m sooooo close!

Hey guys, need some help here trying to figure out how to end my setup.

So far i have a SPA942 connected at my home and my asterisk box is at my office, as some1 else suggested i should open some ports in order to let my IP phone to register itself with the PBX.

Office´s Firewall side:

Inbound TCP port 5060
Inbound UDP port 5060
Inbound UDP ports 5000 - 20000

Home´s firewall side:

None

So far with this configuration i can connect to the remote PBX, recieve and place phonecalls, BUT… i can hear people but they can´t hear me, i know this is prolly a port forwarding stuff, i´m i missing something here?

TY in advance!

What brand/model firewalls do you have at home and office?

Office: D-link DI-624+

Home: Netgear WGR614v6

Both on latest firmwares

I should have asked this earlier; I presume that phones on the same network as the asterisk server and connected to it work fine with 2-way audio?

Well… the phones connected to the office´s network works perfectly, i´m having troubles with the one at home.

BTW just to be sure it´s not the phone it self that is giving me troubles i had also done a couple of tests with X-lite and guess what… same problem, so ya, i guess it´s just a port forwarding i´m missing in my office´s firewall configuration.

I’d look at 2 things.

1- Make sure that the UDP ports you opened up on the firewall match with what asterisk is trying to use.

2- Is your firewall at home capable of handling SIP packets; not all of them are. SIP places the source IP address into the packet and it is up to the firewall to replace that address with its own. If that doesn’t happen, then the asterisk server will decode the packet and get the private address; the RTP packet will not get returned, which will give you one-way audio.

[quote=“acidnecro2”]i can hear people but they can´t hear me, i know this is prolly a port forwarding stuff, i´m i missing something here?[/quote]I had a lot of trouble with one-way audio and it turned out to be codecs. Can’t remember the exact details but it was something like this: I had allow=all so my SIP provider was using something my phones didn’t like, and I got it to work using disallow=all and selectively allowing one after the other till it worked.

Actually b4 i brought this phone to home it was working perfectly at the office, so i don´t think it´s codec related issue… i might be wrong tho.