Open Source Linux VoIP Software - Page 10
Sort By:
VoIP Software for Linux
View 55 business solutions-
Host LLMs in Production With On-Demand GPUsDeploy your model, get an endpoint, pay only for compute time. No GPU provisioning or infrastructure management required.
-
Gemini 3 and 200+ AI Models on One PlatformBuild generative AI apps with Vertex AI. Switch between models without switching platforms.
-
1
This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.
-
2
ictbilling
A client management, billing and e-commerce solution
ICTBilling contains a set of Drupal 7 modules that can be used with Ubercart E-commerce solution and ICTBroadcast, a unified communication auto dialer software. This Client and Billing solution for ITSP has following modules and rules: client module, ictservice module, ratelist module and rules for ubercart. Client module handles clients user interfaces, subscribed service details, service links, list of available addons and current balance. Another module, ICTService creates simple rules that adds top-up, purchase of add-on, assign/revoke permission, create and assign agent / extensions. ICTService also manages a form that allows new customers to choose a package to subscribe. Ratelist module directly connects with service portal's rate list and imports complete rate list in the Drupal table for rate list. -
3
integral
Fast CGI-based AsteriskPBX cluster administration tool.
Allows OnLine monitoring of all kind Asterisk functioality: - Real-Time channel monitoring; - Queue Agents control; - Conference Administration; - Config edition; - Asterisk Core monotoring; -
4
-
Try Google Cloud Risk-Free With $300 in CreditUse your credit across every product. Compute, storage, AI, analytics. When it runs out, 20+ products stay free. You only pay when you choose to.
-
5
jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net -
6
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.
-
7
mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated: https://sourceforge.net/donate/?user_id=2370023 -
8
mumb1e
A Murmur-Server Webinterface written in PHP for Mumble
Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m -
9
-
Catch Bugs Before Your Customers DoMove from alert to fix before users notice. AppSignal monitors errors, performance bottlenecks, host health, and uptime—all from one dashboard. Instant notifications on deployments, anomaly triggers for memory spikes or error surges, and seamless log management. Works out of the box with Rails, Django, Express, Phoenix, Next.js, and dozens more. Starts at $23/month with no hidden fees.
-
10
-
11
nanoagi
Microscopic AGI framework.
Asterisk Gateway Interface often used when it is necessary to build complex IVR system. Nanoagi will help to do it. -
12
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.
-
13
node (nodejs) non-blocking RADIUS client wrapper
-
14
oSKY
Framework implementing SS7 over IP & SIGTRAN interworking and trunking
oSKY (OpenSKY) is a full communication framework to implement PSoTN (Public Social Telephony Network). In contrast with PSTN (Public Switched Telephony Network), oSKY intends to implement an open paradigm in the internet domain, gaining more than 100 years of experience from traditional telephone network. oSKY goals: -enhanced security (SSL, SCTP) -embedded NAT tunneling -Global title discovering on Internet style -Peer2Peer mode and Gateway mode -more to come ... oSKY project drivers: - drastically reduce bandwidth usage by performing smart lookup of enum numbering space. avoiding PSTN outbound trunking when not necessary and linking IPPABXs in the internet domain with failover capabilities. -
15
A JNI wrapper for pjsip. You can use this wrapper to develop Java applications using the pjsip library. At the moment only the pjsua API is implemented. If you would like to obtain a commercial license, or need customisations, please contact us.
-
16
This is a daemon that runs side by side with asterisk. And translates Emails with an XML body into an Asterisk Call with all variables passed from XML as well. Has functionality like an autodialer
-
17
red
Red simulates a lossy and limited network over two ethernet interfaces
-
18
RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,
-
19
Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/
-
20
sippet
C++ SIP stack based on Chrome source code
This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration. -
21
Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
-
22
skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.
-
23
This project implements series of homework assignment from Columbia university course. The main target is to create a SIP enabled thin audio client for Linux.
-
24
-
25
vTiger Freeswitch PBX Integration NYFON
vTiger Freeswitch PBX Integration
vTiger Freeswitch Integration by NYFON. Allows to perform outbound (Click to Call) and incoming (wip) calls from vTiger 6.1+ interface. Modified PBXManager allows to choose between Asterisk and Freeswitch for PBX integration. Same (and extended in the future) functionality as Asterisk interface.
