Transform your applications and workflows into powerful agentic systems at global scale.
Gemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
Get Started Free
Our Free Plans just got better! | Auth0
With up to 25k MAUs and unlimited Okta connections, our Free Plan lets you focus on what you do best—building great apps.
You asked, we delivered! Auth0 is excited to expand our Free and Paid plans to include more options so you can focus on building, deploying, and scaling applications without having to worry about your security. Auth0 now, thank yourself later.
Combination Telephony Engine (presently Asterisk) PBX (in progress FreePBX) and AutoDialer (presently Vicidial) loaded onto Gentoo with detailed instructions. From blank machine to up and running in roughly 3 hours (mostly compile time, by advanced user)
miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.
Stop Cyber Threats with VM-Series Next-Gen Firewall on Azure
Native application identity and user-based security for your Azure cloud
Gain integrated visibility across all traffic in a single pass. Deploy Palo Alto Networks VM-Series to determine application identity and content while automating security policy updates via rich APIs.
Clock in/out from any phone, or from the web portal. Comes with full web portal and reports. Get the latest build from SVN Checkout (Public/SVN Repository). For more information on Asterisk PHP Timeclock, visit http://www.asterisktimeclock.org
The Genalyze log analyzer, a simple yet highly extensible framework for parsing log files, was designed for parsing T-Server log files generated by Genesys contact center software platform, yet is generic and easy enough to adapt for other log formats.
Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
Java Media Library for wiring together Sinks and Sources. Can have rtp Sinks and Sources, FileRecorders/FilePlayers, Microphone, etc. The framework has default sink/sources already. It does not provide Text-to-Speech Source or Speech Recognizer Sink.
Start building on Google Cloud with $300 in free credits. No commitment, no credit card required until you're ready to scale.
Launch your next project with $300 in free Google Cloud credits—no strings attached. Test, build, and deploy without risk. Use your credits across the entire Google Cloud platform to find what works best for your needs. After your credits are used, continue with always-free tier services. Only pay when you're ready to scale. Sign up in minutes and start exploring.
A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
Visual voicemail viewer for Asterisk/AsteriskNOW written in PHP. Users log in with their extension and v/m password and can download messages with the web browser.
Nhabla-Queue: Is a Asterisk report system for inbounds call center campaigns, provide many reports with useful information and graphics for analysis, which can be printed on PDF.
Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.
Nhabla-Bill: Is a report system a billing application inspired on Asterisk-Stat by Areski, offering many useful reports for the analysis, add to the billing function.
J8051 is an API for communication between a 8051 microcontroller and a cell phone using Java ME, Bluetooth and "C" language callback functions, providing a easy way for embed systems developers build a communication between.
RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive
Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.
The Jive API is a lightweight Java framework for writing IVR applications.
The premise for the Jive API is to allow powerful IVR applications to be written without the user having to worry (or know) about the underlying IVR platforms.
SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.
FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.
Development of software and alternative firmware for Asterisk-appliance D-Link HorstBox (DVA-G3342SB), free kubuntu live-dvd for firmware development, sourcecode (GPL) for original D-Link firmware, modifications/patches.
VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).