Sangoma Vega200g gateway with FreePBX

Get FreePBX configured with a Sangoma 200G vega gateway.

This document is an older doc for an old config.  I'm not sure how much it applies now

I'm keeping it here for reference only.  Check this blog for the VEGA100 configuration which is done on JAN of 2021

This installation is specifically with ELASTIX 2.5.0 deployment build using Asterisk and FreePBX

I'm really not going to get into major details of installing this product...but this deployment example is using ELASTIX 2.5.0 - 08 May 2015 build.

Its using a Venga 200g gateway with the following information:

Binary File Name  : VEGA_R101S019
    Release Date      : Mar 16 2016 09:48:13
    Versioning Info   : SIP Firmware Rev 10.01 for H/W Type 15
    Boot Requirements : Boot Loader 04.00

So i pulled the Asterisk side using the following lync:

http://wiki.freepbx.org/display/VG/Freepbx

To sum this up, all you really need to do is create a new SIP trunk

Give it a General Settings Trunk Name ie "SangomaGateway"

Then in OUTGOING SETTING

give it a TRUNK NAME ie "PBX-PRI"

In PEER DETAILS

username=PBX-PRI
secret=as234234234sdfasdfasdfasd3542g4349807
type=friend
qualify=yes
context=from-trunk
insecure=port,invite
host=dynamic

SAVE / APPLY those changes and that is about it.

The "host=dynamic" fixed a bunch of connection issues I had btw.  Its the first time I've ever used that setting.

Aside from creating an outbound route to use that trunk, these were all the changes I made on a base install of ELASTIX.  

VEGA 200G configuration.

I used the configuration for the 200G using the configuration from this site:

https://support.gradwell.com/entries/23304546-How-to-configure-your-Sangoma-Vega-Gateway

It was pretty good.  I've included their work below and noted some changes I had to make to tweak it for the Elastix type of install.

We are assuming you are configuring a brand new unit. Following this procedure will erase/nullify any existing configurations on this box.| Manual configuration



fter logging in, click on Quick Config from the left hand menu.









The quick configuration allows you to set basic entry level values on the unit easily. 

On the Quick Config screen, set the network information that you wish the unit to use, for example the IP address.  Use DHCP or Hardcode the IP as your network needs suit.







Now click on the VoIP tab. Ensure that the values are set as below.

Registration Mode: Off
Outbound Proxy Used?: No
SIP domain: IP OF YOUR ASTERISK BOX
SIP Server IP/Name: IP OF YOUR ASTERISK BOX
Outbound Proxy IP/Name: leave blank
Registration and Authentication ID: leave blank
Authentication Password: leave blank


In the codecs drop down, ensure that g711Alaw64k is selected first, g711Ulaw64k is second and g729 is third.











Now click on E1. In the Telephone number list fields we recommend that you enter .*

NT?  This depends.  Normally YES, check.  Mine, I had to disable it.  So if you encounter issues during turnup of your system, this is a check box that you can come back to.







Now tick New Install? at the top of the screen and then Submit. Ensure that you save the configuration and reboot the system to fully apply the changes.







Once the gateway has rebooted, expand the Expert Config menu on the left hand side and click SIP.









Click on the Modify link that is within the SIP Profiles box.







Look for the From Header 'userinfo' dropdown

Depending on your installation, you might need to use either CALLING PARTY or AUTHENTICATION USERNAME.

I've had issues with inbound.  So this is a spot where you might need to come back to.







If LOCAL DOMAIN is not populated, put in your asterisk IP.

Now scroll to the SIP Proxy entry that is on this page (it's underneath the SIP Profile 1 Proxy Parameters 1 box). Click the Add button and then Modify to enter a new hostname.

IP/DNS Name "Your Asterisk IP", IE 10.4.100.10

Ensure that the Enable box is ticked before clicking Submit. The proxy list should now look like this:



Click the SIP menu link from the main left hand menu.





Scroll to to SIP Authentication Configuration and click SIP Authentication.







Click on Modify.







On this page enter your FreePBX trunk Username and Password.Ensure that Enable is ticked and click Submit.


This should be the USERNAME / PASSWORD you used in your FREEPBX TRUNK screen.








Click the red Save button in the left hand menu and then the red Apply Changes button. Reboot the system and your gateway should be ready to use.

Make sure when you hit SAVE the screen does refresh.  The SAVE should no longer be red.