FreePBX Outbound Problems

So I have a FreePBX server appliance for our office, with Grandstream GXP2160’s around the office. I am setting up our first phone (mine) and I came across a few problems.

The account is registered but when I dial out:
A) I hear no dial tone,
B) When it rings my cell it appears “unknown,” &
C) When I answer both ends will not transfer voice, meaning neither ends can pickup or receive communication (in this case my voice.)

Here is my current set up:


Account Name: Alec Ryncavage
Sip Server:
Secondary Sip Server:
Outbound Proxy:
BLF Server:
SIP user ID: 12005
Authenticate ID: 12005
Authenticate Pass: (My secret password for FreePBX extension menu.)
Name: Alec Ryncavage
VoiceMail User ID:

(Sorry for only one image, I am a new user so they do not allow me to have them.)

Can someone please explain to me what I should do?

***Also just in case this is the problem:

I am using Sip.US as my Sip trunk service and I configured it using there FreePBX client module and I have not change or added any settings to it since I purchased it.

Thank you!

Best regards,
Alec Ryncavage.

I’m curious, if you dial *43 does the echo test work between your phone and the FreePBX Server? Just to rule out any issues between your phone and the FreePBX server directly.

Also your Outbound CallerID, I think, should be configured as "Alec Ryncavage" <XXXXXXXXXX> where the X’s are your phone number that should show up. If you only put your name it would not include the outbound number as this trumps what the outbound number is set to in the trunk. If you set your outbound caller ID on your trunk, then you can leave that field blank in the extension settings. (Someone else correct me if I’m wrong)

There are some things in your post that I hope to clarify (at least, they helped me).

  1. You make is sound like your Asterisk appliance is connected to your Sip.US service. Check the Status Tab and make sure the trunk is registered to your appliance.

  2. Your phone is connected to the appliance through the local LAN connection. Make sure the firewall on the server isn’t getting in your way by turning off the firewall, at least temporarily. The “*43” or “*65” dial codes both send audio back to your phone and verify that you can connect to the Asterisk appliance.

  3. You can hear dialtone, which typically comes from the server. It’s possible that your phone could generate it itself, but that would be unusual. If you want to try another experiment, dial “*12005” and leave yourself a message. That will tell you if your full audio path from and to the appliance are working.

  4. When you dial your cell phone, you aren’t making the call. The appliance is. So, if your cell phone rings when you dial the number, the connection from the appliance to the VOIP provider is up and working.

it sounds like a traditional one-way audio problem with your firewall. In order for your appliance to connect to SIP-US, you need to have access to the Internet accessible from and to the appliance. There are a ton of posts about that out here.

Just learned that for some reason my firewall is stopping two-way audio.

How do I fix this?

See above. Search is your friend.