SIPfoundry / legacy-sipxecsLinks
Unified Communications System
☆87Updated 8 years ago
Alternatives and similar repositories for legacy-sipxecs
Users that are interested in legacy-sipxecs are comparing it to the libraries listed below
Sorting:
- sipXecs Unified Communication System http://www.sipfoundry.org☆26Updated 4 years ago
- SIP Phone Learning Tool☆43Updated 4 years ago
- The Open Source Cloud Communications Platform☆253Updated 2 years ago
- Asterisk module that provides the "Flite" dialplan application, which allows you to use the Flite text to speech engine with Asterisk..Wo…☆28Updated 2 years ago
- The is the central location for the Provisioner Module for VoIP/PBX Servers. Most of the new work is happening inside the v5-dev branch☆124Updated 2 years ago
- Yet Another Telephony Engine - UNOFFICIAL mirror☆115Updated 7 years ago
- Speech recognition script for Asterisk that uses google's speech engine.☆254Updated 7 years ago
- A2Billing is a telecom switch and billing system capable of providing and billing a range of telecom products and services to customers s…☆189Updated 3 years ago
- The original single-server, multi-tenant, switch agnostic UI built by 2600Hz☆105Updated 10 years ago
- First Open Source Billing Platform for FreeSWITCH☆49Updated 12 years ago
- Free and convenient server process for routing SMS text messages between your applications and SMPP gateways. Interacts with your applica…☆52Updated 9 years ago
- XMPP MUC bot that interacts with Asterisk☆13Updated 9 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆172Updated 5 months ago
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆305Updated 3 years ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆173Updated 8 months ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆41Updated 11 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆83Updated 9 years ago
- Unified Communications System☆36Updated last year
- SMS, Fax, Voice Broadcasting and auto dialer Software, A unified communications open source autodialer developed over freeswitch communic…☆95Updated 11 months ago
- ☆88Updated 5 years ago
- Sylk WebRTC client☆62Updated 2 weeks ago
- Asterisk manager interface (ami) client for nodejs☆102Updated last year
- SIP Express Media Server, very fast and flexible SIP media server☆70Updated last week
- Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)☆175Updated this week
- Monster UI Application: Simplified PBX configuration for end-users☆31Updated last month
- Javascript library to build a web-broswer softphone☆103Updated 6 months ago
- Real time web based Asterisk monitoring with Meteor☆27Updated last week
- TESTING - replicated hourly from Google Code SVN using sync2git☆58Updated 10 years ago
- Restcomm Session Border Controller☆24Updated 6 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆82Updated 11 years ago