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Connect a mPBX SIP Trunk to an Asterisk PBX

Asterisk, FreePBX, Vicidial, Elastix mPBX SIP Trunk Configuration

If you are using a Asterisk onsite or hosted PBX and require to connect a mPBX SIP trunk follow the guide below;

The first stage is to login to your customer portal (customer.voipline.net.au) and click on the order services screen, add a new SIP trunk and the required amount of channel you would like to assign to this SIP trunk. Also make sure that you choose a call plan that suits your outbound call volumes. 

Once selected the next process is to submit the order and click on the mPBX tab at the top of the customer portal. Once inside the call flow drag a SIP trunk object onto the screen from the left hand tool box and click the settings cog to configure the SIP trunk object and locate your registration details. 

The next process is to click the drop down menu labelled mode and select create SIP registration, this will then generate you your SIP trunk details that you will need to enter into your sip.conf configuration file. Make sure in mode options that you select the call plan and all the codec that you would like to use. The final stage before attempting to register is to save the SIP trunk configuration inside mPBX and press the apply configuration button on the left hand side below the toolbox. Important note! if you do not apply the configuration and attempt to register the SIP trunk the registration will fail and this may result in your IP address being blocked.

Now you are ready to configure your Asterisk PBX to connect to the VoIPLine network and start making and receiving calls. First, open the sip.conf file of your Asterisk based PBX and enter in the following trunk configuration details and registration string if required. 

Peer Details:

[voipline telecom sip trunk]
username=[SIP username]
type=peer
secret=[YOURPASS]
qualify=yes
nat=yes
host=sipm.voipline.net.au (or 103.242.182.110 depending on if you have DNS configured in your PBX)
port=7060
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g722
allow=ulaw
insecure=port,invite


Register String:

register => sipusername:sippassword@sipm.voipline.net.au:7060

The final stage is to add all of your DID's to your SIP trunk inside your Asterisk PBX configuration files or GUI and create your inbound and outbound routes, test calls and if you require any further assistance contact support@voipline.net.au. 

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