[SOLVED] Please point me in the right direction. I just purchased a SIPstation line and DID as well as Yealink T46G

Sangoma Trunking
1

So far I have set up my IP phone with a static IP address and I have created an extension. I can login to my ip phone, freePBX local server, and my SIPstation account. When I call my DID number it doesn’t ring and goes immediately to the prompt, “The person at extension XXXX is unavailable please leave a message after the tone.”

I am using a T46G IP Phone as well as SIPstation.

I have tried to use the documentation but because I am new I am uncertain with what I am doing wrong. Please provide me with a troubleshooting strategy. I would really love to dig into the IVR after this.

Thank you all so much,

Phil

2

Have you created an inbound route for your DID? You need to do that.

3

Okay so now it rings but there is no audio. I pretty sure I have a NAT settings problem now.

4

Is your firewall setup to allow ports 10000-20000 UDP to your PBX. Also in the SIP Settings module in FreePBX make sure your external IP address is setup and your local subnets are defined.

5

First off thank you so much for replying.

tonyclewis,

I have set up my FreePBX with my external IP address and local subnets.

I’ve attempted to look at the sip_nat.conf via PuTTy however nothing was in the file.

Looking at Connectivity >> SIPstation

Looking at Settings >> Asterisk Settings

Looking at Settings >> Asterisk Settings >> then clicking Chan SIP at the top right

I’ll go ahead and double check my firewall configurations as you suggested as well.

6

$%#^ING SUCCESS!!!

You all are truly stand up guys!

Thank you so much.

I’m sure I’ll be on here in a couple of days.

:slight_smile:

7

FYI I solved this by configuring my modem via Comcast Business Gateway. Per tonyclewis I configured my ports for each of my ip devices involved in my VOIP infrastructure.