Search Results for "command line client"
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Showing 22 open source projects for "command line client"
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Vibes don’t ship, Retool doesVibe coding tools create cool demos, but Retool helps you build software your company can actually use. Generate internal apps that connect directly to your data—deployed in your cloud with enterprise security from day one. Build dashboards, admin panels, and workflows with granular permissions already in place. Stop prototyping and ship on a platform that actually passes security review.
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Atera all-in-one platform IT management software with AI agentsAtera’s AI agents don’t just assist, they act. From detection to resolution, they handle incidents and requests instantly, taking your IT management from automated to autonomous.
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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Callflow Sequence Diagram Generator
Callflow Sequence Diagram Generator
The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic -
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PolycomVVXControl
A command line utility for remote controlling Polycom VVX IP phones
Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file. -
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SIP Data Filter (SiDaFir)
Simple and efficient tool for SIP trace filtering
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic... -
Grafana: The open and composable observability platformGrafana is the open source analytics & monitoring solution for every database.
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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nanoagi
Microscopic AGI framework.
Asterisk Gateway Interface often used when it is necessary to build complex IVR system. Nanoagi will help to do it. -
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for... -
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SMPTE 2022-1 library for VLC & co
Optimized and cross platform SMPTE 2022 FEC library in C, Python, Java
Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib -
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VoIPGMap
Graph active VoIP calls on Google Maps.
A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map. -
Leverage AI to Automate Medical CodingAs a healthcare provider, you should be paid promptly for the services you provide to patients. Slow, inefficient, and error-prone manual coding keeps you from the financial peace you deserve. XpertDox’s autonomous coding solution accelerates the revenue cycle so you can focus on providing great healthcare.
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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
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What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.
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SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.
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VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K
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Lwazi is a robust telephony platform aiming to facilitate speedy development of experimental applications without sacrificing power by combining Asterisk with the MobilIVR Python interface bundled into a single build with a unified control interface.
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A set of scripts to allow sending and receiving of fax documents on HylaFAX using res_fax and res_fax_digium of Asterisk
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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.
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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.
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A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.
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SMSBuster has been designed for SMS sending from the command line over VoipBuster. Perfect if you need to send a SMS from your own scripts. It is very easy to use and does not require any extra libraries.Works perfect with Nagios for alerting !
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Acv is a PHP program which allows you to monitor Asterisk servers by viewing active calls, channels, and overall server performance numbers.
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This is a SIP based conference server. Allow to host multiple conference bridges and allow multiple participants in each conference.
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