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* A computer or a VM if you're not containerizing it | * A computer or a VM if you're not containerizing it | ||
* a trunk account and extenstion for said trunk | * a trunk account and extenstion for said trunk | ||
*(optional) a SIP phone (available in multiple flavours from the space) | *(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 | *(optional) a Softphone like Linphone,3CXPhone,SimpleSIP etc etc | ||
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
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