FreePBX (PJSIP)

= Prerequisites =
== Sub account creation ==
Before configuring your FreePBX Trunk, you will need to have a sub-account ready to be connected. 
If you don’t have a sub-account yet, you need to create a new one. 
To do so, Log into your VoIP.ms portal, navigate to “Sub-Account” and click on “Create a sub-account”.
Your sub-account needs to have the minimum required in order to work with FreePBX.
: -	Username 
: -	Password 
: -	Device Type: (Asterisk, IP PBX, Gateway, VoIP Switch)
: -	CallerID Number: (I use a system capable of passing its own CallerID)

== Set a DID for Incoming calls ==
In order to received calls from a specific DID and leverage all your PBX functionalities, you needs to route directly your DID to your sub-accounts.
To do so, navigate through the navigation bar and go under “DID Numbers” then “Manage DID”.
To edit a DID, you will need to edit this DID by clicking on the yellow button and apply the proper SIP/IAX account.
Note or select your preferred PoP Server; your DID Numbers need to reflect the same PoP server in order to receive incoming calls. 

= Configuration of FreePBX =
== Creating a new trunk ==
On your FreePBX panel, Click on the menu [Connectivity] menu, then [Trunk]. When you are on the trunk page, Click on [+ Add Trunk] and select [+ Add SIP (Chan_pjsip) Trunk].

[[File:FreePBX_PJSIP_Trunk_ADD-PJSIP.png|750px]]


=== General Tab ===
: Trunk Name: This is only to identify your trunk for your own purposes.
: Outbound CallerID: '''"CALLERID NAME" <##########>''' (your 10 digit DID # without any dot)
: Note: The name should be '''max 15 characters''', Must be in '''CAPITAL LETTERS''' without specials characters. Spaces are allowed.

[[File:FreePBX_PJSIP_Trunk_ADD-PJSIP-general.png|750px]]


=== pjsip Settings ===
==== General ====
: Username: ######_username (This will be your VoIP.ms sub-account)
: Secret: This will be the password of your sub-account.
: SIP Server: Your preferred PoP Server.
: SIP Server Port: 5060 or one of our alternative port 5080 or 42872


[[File:FreePBX_PJSIP_Trunk_ADD-PJSIP_pjsipSettings-General.png|750px]]

==== Advanced ====
: From domain: Specify your WAN IP address for your PBX. 

== Inbound Route Setup ==
In order to route your DID in your PBX you will need to create a inbound route. To do so, Go to the menu [Connectivity] then [Inbound Route], on this page, click on [Add Inbound Route]
: Description: You can add a description for your own purposes.
: DID Number: This will be your 10 digits DID that you have routed to your sub-account. 

[[File:FreePBX_PJSIP_Inbound_ADD-PJSIP.png|750px]]


== Outbound Routes ==
=== Route settings ===
: Route name: Give your outbound route a name for your own purposes. <br/>
: Route CID: '''"CALLERID NAME" <##########>''' (your 10 digit DID # without any dot).
: Note: The name should be '''max 15 characters''', Must be in '''CAPITAL LETTERS''' without specials characters. Spaces are allowed. If a CID in your trunk has not been forced, you need to specify the outbound caller ID used with this outbound route. 
: Trunk sequence for Matched routes: You will need to the new VoIP.ms trunk you have created. 

[[File:FreePBX_PJSIP_Outbound_ADD-PJSIP.png|750px]]

=== Dial Patterns ===

== Outbound Caller ID Name for call to Canada ==
Since Canada’s provider are not using the CNAM database to pass the caller ID Name, you will need to provide this information directly from your trunk. The way to display a name and a number in a trunk is: "CALLERID NAME" <##########>.

<div style="border-left: 6px solid blue; background-color: lightgrey; width: 85%;">
 '''IMPORTANT''': 
   - We suggest entering your outbound Caller ID Name must be in '''CAPITAL LETTERS'''. This will appears more clearly/visible on some devices.
   - You must NOT use any special characters, they will not be displayed. 
   - Do not exceed '''15 characters''' max! Some of regular Canadian providers will not show more than '''15 characters'''. We suggest shrinking or adapt your caller ID. 
   - Spaces are allowed in a caller id name.
</div>

For USA destinations, you will need to request a CNAM Database update through the live chat or by submitting a ticket request. Note that the '''same criteria must be respected'''.





''This article was made in collaboration with Chris Sherwood from Crosstalk Solution. You can find his Youtube explanatory video about PJSIP Trunk here : https://www.youtube.com/watch?v=Mu1OxktwURg
and his blog article here : https://crosstalksolutions.com/voip-ms-setup-using-pjsip-on-freepbx/''