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Merge pull request #1297 from AlessioCasco/homer

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@ -365,6 +365,7 @@ _[SIP](https://en.wikipedia.org/wiki/Session_Initiation_Protocol) telephony soft
- [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 & 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`