mirror of
				https://github.com/awesome-selfhosted/awesome-selfhosted.git
				synced 2025-11-03 20:45:25 +01:00 
			
		
		
		
	grouping SIP and IPBX sections
This commit is contained in:
		
							
								
								
									
										22
									
								
								README.md
									
									
									
									
									
								
							
							
						
						
									
										22
									
								
								README.md
									
									
									
									
									
								
							@@ -26,8 +26,7 @@ See [Contributing](.github/CONTRIBUTING.md).
 | 
				
			|||||||
      - [Mailing lists and newsletters](#mailing-lists-and-newsletters)
 | 
					      - [Mailing lists and newsletters](#mailing-lists-and-newsletters)
 | 
				
			||||||
      - [Webmail clients](#webmail-clients)
 | 
					      - [Webmail clients](#webmail-clients)
 | 
				
			||||||
    - [IRC](#irc)
 | 
					    - [IRC](#irc)
 | 
				
			||||||
    - [SIP](#sip)
 | 
					    - [SIP/IPBX](#sip)
 | 
				
			||||||
    - [IPBX](#ipbx)
 | 
					 | 
				
			||||||
    - [Social Networks and Forums](#social-networks-and-forums)
 | 
					    - [Social Networks and Forums](#social-networks-and-forums)
 | 
				
			||||||
    - [XMPP](#xmpp)
 | 
					    - [XMPP](#xmpp)
 | 
				
			||||||
      - [XMPP Servers](#xmpp-servers)
 | 
					      - [XMPP Servers](#xmpp-servers)
 | 
				
			||||||
@@ -389,25 +388,18 @@ _[IRC](https://en.wikipedia.org/wiki/Internet_Relay_Chat) communication software
 | 
				
			|||||||
 | 
					
 | 
				
			||||||
**[`^        back to top        ^`](#)**
 | 
					**[`^        back to top        ^`](#)**
 | 
				
			||||||
 | 
					
 | 
				
			||||||
_[SIP](https://en.wikipedia.org/wiki/Session_Initiation_Protocol) telephony software_
 | 
					_[SIP](https://en.wikipedia.org/wiki/Session_Initiation_Protocol)/[IPBX](https://en.wikipedia.org/wiki/IP_PBX) telephony software_
 | 
				
			||||||
 | 
					
 | 
				
			||||||
- [Asterisk](http://www.asterisk.org/) - Easy to use but advanced IP PBX system, VoIP gateway and conference server. `GPL-2.0` `C`
 | 
					- [Asterisk](http://www.asterisk.org/) - Easy to use but advanced IP PBX system, VoIP gateway and conference server. `GPL-2.0` `C`
 | 
				
			||||||
- [FreeSWITCH](https://freeswitch.org/) - Scalable open source cross-platform telephony platform. ([Source Code](https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse)) `MPL-2.0` `C`
 | 
					 | 
				
			||||||
- [Homer](https://www.sipcapture.org/) - Troubleshooting and monitoring VoIP calls. ([Source Code](https://github.com/sipcapture/homer)) `AGPL-3.0` `Angular/C`
 | 
					 | 
				
			||||||
- [Kamailio](http://www.kamailio.org/w/) - Modular SIP server (registrar/proxy/router/etc). ([Source Code](https://github.com/kamailio/kamailio)) `GPL-2.0` `C`
 | 
					 | 
				
			||||||
- [Ostel](https://dev.guardianproject.info/projects/ostel/wiki/Server_Documentation) - Secure SIP telephony setup with ZRTP encryption. `GPL-3.0` `Ruby`
 | 
					 | 
				
			||||||
- [Tapir](http://www.sip3.io/) - Troubleshooting and real-time monitoring of VoIP-based systems. ([Source Code](https://github.com/sip3io/tapir)) `Apache-2.0` `Java/Kotlin`
 | 
					 | 
				
			||||||
 | 
					 | 
				
			||||||
### IPBX
 | 
					 | 
				
			||||||
 | 
					 | 
				
			||||||
**[`^        back to top        ^`](#)**
 | 
					 | 
				
			||||||
 | 
					 | 
				
			||||||
_[IPBX](https://en.wikipedia.org/wiki/IP_PBX) telephony software_
 | 
					 | 
				
			||||||
 | 
					 | 
				
			||||||
- [ASTPP](https://www.astppbilling.org/) - is an Open Source VoIP Billing Solution for Freeswitch. It supports prepaid and postpaid billing with call rating and credit control. It also provides many other features. ([Source Code](https://github.com/iNextrix/ASTPP)) `AGPL-3.0` `PHP`
 | 
					- [ASTPP](https://www.astppbilling.org/) - is an Open Source VoIP Billing Solution for Freeswitch. It supports prepaid and postpaid billing with call rating and credit control. It also provides many other features. ([Source Code](https://github.com/iNextrix/ASTPP)) `AGPL-3.0` `PHP`
 | 
				
			||||||
- [Freepbx](http://www.freepbx.org) - Web-based open source GUI that controls and manages Asterisk. ([Source Code](http://git.freepbx.org/projects/FREEPBX)) `GPL-2.0` `PHP`
 | 
					- [Freepbx](http://www.freepbx.org) - Web-based open source GUI that controls and manages Asterisk. ([Source Code](http://git.freepbx.org/projects/FREEPBX)) `GPL-2.0` `PHP`
 | 
				
			||||||
 | 
					- [FreeSWITCH](https://freeswitch.org/) - Scalable open source cross-platform telephony platform. ([Source Code](https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse)) `MPL-2.0` `C`
 | 
				
			||||||
- [FusionPBX](https://www.fusionpbx.com/) - Open source project that provides a customizable and flexible web interface to the very powerful and highly scalable multi-platform voice switch called FreeSWITCH. ([Source Code](https://github.com/fusionpbx/fusionpbx)) `MPL-1.1` `PHP`
 | 
					- [FusionPBX](https://www.fusionpbx.com/) - Open source project that provides a customizable and flexible web interface to the very powerful and highly scalable multi-platform voice switch called FreeSWITCH. ([Source Code](https://github.com/fusionpbx/fusionpbx)) `MPL-1.1` `PHP`
 | 
				
			||||||
 | 
					- [Homer](https://www.sipcapture.org/) - Troubleshooting and monitoring VoIP calls. ([Source Code](https://github.com/sipcapture/homer)) `AGPL-3.0` `Angular/C`
 | 
				
			||||||
 | 
					- [Kamailio](http://www.kamailio.org/w/) - Modular SIP server (registrar/proxy/router/etc). ([Source Code](https://github.com/kamailio/kamailio)) `GPL-2.0` `C`
 | 
				
			||||||
- [Kazoo](http://2600hz.org/) - KAZOO is an open-source, highly scalable software platform designed to provide carrier-grade VoIP switch functions and features. ([Source Code](https://github.com/2600hz/KAZOO)) `MPL-1.1` `Erlang`
 | 
					- [Kazoo](http://2600hz.org/) - KAZOO is an open-source, highly scalable software platform designed to provide carrier-grade VoIP switch functions and features. ([Source Code](https://github.com/2600hz/KAZOO)) `MPL-1.1` `Erlang`
 | 
				
			||||||
 | 
					- [Ostel](https://dev.guardianproject.info/projects/ostel/wiki/Server_Documentation) - Secure SIP telephony setup with ZRTP encryption. `GPL-3.0` `Ruby`
 | 
				
			||||||
 | 
					- [Tapir](http://www.sip3.io/) - Troubleshooting and real-time monitoring of VoIP-based systems. ([Source Code](https://github.com/sip3io/tapir)) `Apache-2.0` `Java/Kotlin`
 | 
				
			||||||
- [Wazo](http://wazo.community/) - Full-featured IPBX solution built atop Asterisk with integrated Web administration interface and REST-ful API. ([Source Code](https://github.com/wazo-pbx)) `GPL-3.0` `Python/PHP`
 | 
					- [Wazo](http://wazo.community/) - Full-featured IPBX solution built atop Asterisk with integrated Web administration interface and REST-ful API. ([Source Code](https://github.com/wazo-pbx)) `GPL-3.0` `Python/PHP`
 | 
				
			||||||
 | 
					
 | 
				
			||||||
### Social Networks and Forums
 | 
					### Social Networks and Forums
 | 
				
			||||||
 
 | 
				
			|||||||
		Reference in New Issue
	
	Block a user