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= Introduction = | = Introduction = | ||
FreePBX is a GUI for Asterisk, currently maintained by the same company behind Asterisk Sangoma, | |||
it's available as a standalone OS based on CentOS, a community made Docker container, or installed next to asterisk on your preffered OS. | |||
= Requirements = | = Requirements = | ||
The requirements are as follow | |||
* A Wireguard or other VPN connection to the space | |||
* A computer or a VM if you're not containerizing it | |||
* a trunk account and extenstion for said trunk | |||
*(optional) a SIP phone (available in multiple flavours from the space recommended from the chef is [https://nurdspace.nl/Asterisk/Polycom_VVX_Series Poly(com) VVX series]) | |||
*(optional) a Softphone like Linphone,3CXPhone,SimpleSIP etc etc | |||
= Demo Setup = | = Demo Setup = | ||
At home i run it as follows | |||
*A VM on a "server" using the FreePBX OS iso [https://www.freepbx.org/downloads/ link] | |||
*Within said OS a Wireguard tunnel to the space [https://www.wireguard.com/compilation/ link] (you might need Google to help you out due it throwing some errors) | |||
*A Polycom/Poly VVX400 series phone | |||
== Trunks == | |||
Found under Connectivity>Trunks | |||
=== General === | |||
; | |||
; Trunk Name: Custom Name | |||
; Hide CallerID: No | |||
; Outbound CallerID: Example 3011 (Tahtkev) | |||
; CID Options: Allow Any CID | |||
; Maximum Channels: N/A | |||
; Asterisk Trunk Dial Options: System | |||
; Continue if Busy: No | |||
; Disable Trunk: No | |||
; Monitor Trunk Failures: No | |||
=== Dialed Number Manipulation Rules === | |||
We do nothing here we'll manipulate that within Outbound Routes | |||
=== pjsip Settings === | |||
==== General ==== | |||
Username: "your trunk username" | |||
Auth username: ^ | |||
Secret: your trunk password that you've recieved | |||
Authentication: Outbound | |||
Registration: Send | |||
Language Code: English | |||
SIP Server: 10.208.11.13 | |||
SIP Server Port: 5060 | |||
Context: from-internal | |||
Transport: 0.0.0.0-udp | |||
==== Advanced ==== | |||
DTMF MODE: RFC4733 | |||
= Verification that things are working = | = Verification that things are working = | ||
Lorem ipsum dolor sit amet, consectetur adipiscing elit. Vivamus tristique malesuada arcu. Fusce vestibulum ut mauris nec mollis. Quisque lectus dui, ultrices eget arcu ut, porttitor laoreet ipsum. Morbi quam erat, cursus ac dui vel, facilisis consectetur ante. Integer eleifend congue posuere. Fusce augue turpis, luctus sit amet ultrices eget, elementum ut augue. Quisque eu finibus eros. Sed accumsan pretium ultrices. Etiam nec maximus tortor, eu semper risus. Aliquam erat volutpat. Nam pretium ac nisl nec maximus. Vivamus pellentesque euismod volutpat. Phasellus risus purus, vehicula in elementum ut, ultricies a lectus. | Lorem ipsum dolor sit amet, consectetur adipiscing elit. Vivamus tristique malesuada arcu. Fusce vestibulum ut mauris nec mollis. Quisque lectus dui, ultrices eget arcu ut, porttitor laoreet ipsum. Morbi quam erat, cursus ac dui vel, facilisis consectetur ante. Integer eleifend congue posuere. Fusce augue turpis, luctus sit amet ultrices eget, elementum ut augue. Quisque eu finibus eros. Sed accumsan pretium ultrices. Etiam nec maximus tortor, eu semper risus. Aliquam erat volutpat. Nam pretium ac nisl nec maximus. Vivamus pellentesque euismod volutpat. Phasellus risus purus, vehicula in elementum ut, ultricies a lectus. | ||
Latest revision as of 19:15, 1 November 2022
Introduction
FreePBX is a GUI for Asterisk, currently maintained by the same company behind Asterisk Sangoma, it's available as a standalone OS based on CentOS, a community made Docker container, or installed next to asterisk on your preffered OS.
Requirements
The requirements are as follow
- A Wireguard or other VPN connection to the space
- A computer or a VM if you're not containerizing it
- a trunk account and extenstion for said trunk
- (optional) a SIP phone (available in multiple flavours from the space recommended from the chef is Poly(com) VVX series)
- (optional) a Softphone like Linphone,3CXPhone,SimpleSIP etc etc
Demo Setup
At home i run it as follows
- A VM on a "server" using the FreePBX OS iso link
- Within said OS a Wireguard tunnel to the space link (you might need Google to help you out due it throwing some errors)
- A Polycom/Poly VVX400 series phone
Trunks
Found under Connectivity>Trunks
=== General === ; ; Trunk Name: Custom Name ; Hide CallerID: No ; Outbound CallerID: Example 3011 (Tahtkev) ; CID Options: Allow Any CID ; Maximum Channels: N/A ; Asterisk Trunk Dial Options: System ; Continue if Busy: No ; Disable Trunk: No ; Monitor Trunk Failures: No
Dialed Number Manipulation Rules
We do nothing here we'll manipulate that within Outbound Routes
pjsip Settings
General
Username: "your trunk username" Auth username: ^ Secret: your trunk password that you've recieved Authentication: Outbound Registration: Send Language Code: English SIP Server: 10.208.11.13 SIP Server Port: 5060 Context: from-internal Transport: 0.0.0.0-udp
Advanced
DTMF MODE: RFC4733
Verification that things are working
Lorem ipsum dolor sit amet, consectetur adipiscing elit. Vivamus tristique malesuada arcu. Fusce vestibulum ut mauris nec mollis. Quisque lectus dui, ultrices eget arcu ut, porttitor laoreet ipsum. Morbi quam erat, cursus ac dui vel, facilisis consectetur ante. Integer eleifend congue posuere. Fusce augue turpis, luctus sit amet ultrices eget, elementum ut augue. Quisque eu finibus eros. Sed accumsan pretium ultrices. Etiam nec maximus tortor, eu semper risus. Aliquam erat volutpat. Nam pretium ac nisl nec maximus. Vivamus pellentesque euismod volutpat. Phasellus risus purus, vehicula in elementum ut, ultricies a lectus.