Sangoma Vega 100g Configuration with FREEPBX

 

This document will discuss how to create a SIP or PJSIP trunk between a Vega 100 and an Asterisk box.  We'll be using FreePBX distribution.

This was built on the VEGA Firmware Version VEGA_R111S033

FreePBX version 14.0.16.4

The process will be first to build the TRUNK in the Asterisk FREPBX config

Then build the trunk connection in the Vega100 device

In the examples used

• VEGA box will be 10.1.105.50
• FreePBX will be 10.1.105.52
• SIP Trunk will be VegaSIP
• PJSIP Trunk will be VegaPJSIP

SOME other info.

You can console to the port using a serial port (cisco cable works) and the baud rate is 115200

If you get stuck on the configs and want to start again, use "FACTORY RESET" to reset the configs.  This won't reset your LAN or password settings, just configs.   once issued, follow it up with a "SAVE" followed by a "REBOOT" 

Also in console (such as putty) "CTRL+H" is backspace if you make a syntax typo

CONFIGURATION STEPS

Asterisk FREEPBX Trunk configuration

Currently in the FreePBX gui, you can have SIP or PJSIP trunks.  You can also have your configuration setup so that one or the other is simply disabled

Either will work, just different configuration depending on your situation, all 3 will be discussed.

SIP TRUNK

Create a SIP trunk with the Trunk name of "VegaSIP"

Make the "Trunk Names" and the username match.  Put this in the Outgoing PEER details box

username=VegaSIP
type=peer
trustrpid=yes
sendrpid=yes
secret=VegaSIPpass
qualify=yes
insecure=port,invite
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-internal
allow=ulaw,alaw

Save and apply this.

PJSIP TRUNK

Trunk Name (aka Username) = VegaSIP

Secret = VegaSIPpass

Authentication = INBOUND

REGISTARTION = RECEIVE

Vega 100G Configuration

STEP 1

Login to your Vega 100 gui  (http://IPof Vega )

Click on QUICK CONNECT

Set your Country

Click on VOIP

At point of writing this, I'm not sure what "VOIP" section in quick config does exactly.  because it doesn't seem to populate anything in the main configuration.  But I'll include this step anyway till I can figure it out.

Put in the IP of your asterisk PBX server.  No registration mode required.

The above entry reflects in this entries

   set .quick.voip.proxy.auth_name="TEST"

   set .quick.voip.proxy.auth_pwd="MARKET"

   set .quick.voip.proxy.outbound_profile_sig_transport="udp"

   set .quick.voip.proxy.outbound_proxy_port="5160"

   set .quick.voip.proxy.proxy_addr="10.1.105.52"

   set .quick.voip.proxy.tls_port="5061"

(This document won't reflect the T1/E1 configurations)

Click on “NEW INSTALL” check box

Click on “SUBMIT”

REBOOT

STEP 2

Goto EXPERT CONFIG -> SIP ->  "MODIFY" (under "SIP PROFILES")

Set LOCAL DOMAIN - IP of your PBX

Click SUBMIT

STEP 3

Below "SIP Profile 1 Proxy Parameters 1"

Select the "Chg? MODIFY" button 

Put in your IP of your PBX and ensure its "ENABLED"

Select SUBMIT

STEP 4

Scroll down to the bottom in the section with "SIP Registrar" and select "MODIFY"

Ensure it is enabled

Put in the IP of your PBX and click on SUBMIT

STEP 5

Go to Expert Config -SIP

In the "Registration" section

Enable Registration check box 

Press submit

STEP 6

Go to EXPERT CONFIG - SIP - "SIP Authentication Configuration"

Click on the "SIP Authentication" link

Select MODIFY

Enter in your Asterisk FreePBX trunk use name and password

Select "ENABLE"

Press SUBMIT

STEP 7

Goto EXPERT CONFIG - SIP 

Select "SIP Registration Users Configuration"

You should now see a profile that reflects the user name of your trunk.  Click on MODIFY beside the one with your trunk user name

Click on MODIFY beside the entry with an authentication name

Put in the USER name, set to enable, and press submit

click on SAVE

click on APPLY CHANGES

Reboot unit

If everything is configured, in your asterisk log you should see something like this:

Contact VegaSIP/sip:01@10.9.105.15:5060 is now Reachable.  RTT: 10.895 msec

And in the SANGOMA gui you'll see something like this: