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
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: