Update Languages in SIP

README.md

@@ -511,7 +511,7 @@ Simple deployment of [E-mail](https://en.wikipedia.org/wiki/Email) servers, e.g.
 
 - [Asterisk](https://www.asterisk.org/) - Easy to use but advanced IP PBX system, VoIP gateway and conference server. `GPL-2.0` `C`
 - [ASTPP](https://www.astppbilling.org/) - 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`
-- [Eqivo](https://eqivo.org/) - Eqivo implements an API layer on top of FreeSWITCH facilitating integration between web applications and voice/video-enabled endpoints such as traditional phone lines (PSTN), VoIP phones, webRTC clients etc. ([Source Code](https://github.com/rtckit/eqivo)) `MIT` `PHP`
+- [Eqivo](https://eqivo.org/) - Eqivo implements an API layer on top of FreeSWITCH facilitating integration between web applications and voice/video-enabled endpoints such as traditional phone lines (PSTN), VoIP phones, webRTC clients etc. ([Source Code](https://github.com/rtckit/eqivo)) `MIT` `Docker/PHP`
 - [Flexisip](https://www.linphone.org/technical-corner/flexisip/) - A complete, modular and scalable SIP server, includes a push gateway, to deliver SIP incoming calls or text messages on mobile device platforms where push notifications are required to receive information when the app is not active in the foreground. ([Source Code](https://github.com/BelledonneCommunications/flexisip)) `AGPL-3.0` `C/Docker`
 - [Freepbx](https://www.freepbx.org) - Web-based open source GUI that controls and manages Asterisk. ([Source Code](https://git.freepbx.org/projects/FREEPBX)) `GPL-2.0` `PHP`
 - [FreeSWITCH](https://freeswitch.org/) - Scalable open source cross-platform telephony platform. ([Source Code](https://github.com/signalwire/freeswitch)) `MPL-2.0` `C`