Search Results for "asterisk-java"

...This agent utilizes embedded artificial intelligence models operating directly on the softphone, ensuring efficiency and reduced operational costs. This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once.

...Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations (CRM, call recording, automation) Plugin Marketplace: Community and premium extensions Headset Support: Jabra, Poly, Sennheiser, Logitech with HID controls Call Quality: Real-time MOS Score, Jitter, Packet Loss and RTT monitoring Enterprise: Silent MSI installer, GPO support, JSON centralized config Compatibility: 3CX, Asterisk, FreePBX, OVH, Twilio, RingCentral, and more Language packages : https://sipline.feelautom.fr/languages

The Open Source CRM software SuiteCRM is enhanced to have communications capabilities. SuiteCRM is customized and integrated with ICTContact, the Asterisk-based advanced contact center software. It enables CRM users to make and receive calls directly from CRM, also export leads/contacts to the contact center, and run campaigns through ICTContact to connect with their customers with ease. ICTContact is advanced contact center software featuring omnichannel communications, unified communications, IVR studio, CRM operations, Collaboration, CRM business automation and integration, and multi-tenant and white label support suitable for contact centers, call centers, entrepreneurs, and Service Providers. ...

...It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure telephone system for your company, save, take advantage of recent innovations and provide opportunities to integrate your business processes if you wish to do so.

jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP.

...Many VoIP distributors use the best of each VoIP server, termination to be able to be more competitive and to have better price and quality. Magnus Sever is softphone integration service installed in LINUX featuring program like: asterisk, freepbx, a2billing, dahdi among others. It allows integration of how many terminations as you wish, the best second track and the best price for each prefix.

The Bria Desktop API has moved to github at https://github.com/CounterPathAPI. Additional samples for Java and Javascript can be found there. The Bria Desktop allows third-party applications to control Bria for Windows softphone clients. By leveraging the Application API, third-party applications can perform commands such as starting an audio or video call, answering a call or placing a call on hold. CounterPath is now encouraging third-party-application developers to integrate their applications with Bria. ...

camerubuntu-phone-12.04 est un système ubuntu intégrant la téléphonie IP. Avec ce système Linux, vous disposerez d'un serveur de téléphonie IP Asterisk orienté web qui vous permettra d'effectuer sur votre réseau des appels téléphoniques par ordinateurs, par téléphones IP et mếme en utilisant vos smartphones. Une fois installé, il suffit juste de connecter le PC au réseau, de créer les numéros en utilisant un navigateur, installer un client softphone comme 3CX ou autre dans les PC clients et vous pourrez passer des appels dans votre réseau de façon illimitée.

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

IMS-Communicator is an IMS softphone based on the old version of the sip-communicator java project, implemented on top of the JAIN-SIP stack and the Java Media Framework API. It can be used with Open IMS Core (http://www.openimscore.org).

TIPcon1 is a standards-compliant soft phone using text, voice and video with accessibility features.

This project is a SIP based softphone developed as a Java applet that can be embedded into a website. The applet has a simple user interface and is intended to be used by journalists to make and deposit audio recordings via a SIP proxy.

Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

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).

Among the functionality: Include a large variety of codecs (G711, GSM, and SPEEX) - Protocol SIP - Other technical functionalities the support of DTMF (tonalities) although support ENUM (to employ numbers of SIP instead of the addresses of SIP).

HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت

German Stundent Projekt - Java VoIP Softphone

Softphone SIP com voz, vídeo e Mensagem Instantânea. DTMF, Nat-Traversal, Suporte a Skin(Macromedia Flash), SMS, Historico de Chamadas, Presença, Hold, Auto-Answer e etc. Sip Softphone with Voice, Video and IM.

SIP SoftPhone desenvolvido em Java baseado na JAIN-SIP, JMF e Sip-Communicator 1.0. Possui também o envio de SMS, Agenda, NAT Traversal, DTMF Send/Receive, CallTo e etc. A Java SIP SoftPhone based in JAIN-SIP, JMF and Sip-Comminicator 1.0.

The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit