My online writings. This time its about OpenSIPS 101
☆28Mar 20, 2015Updated 11 years ago
Alternatives and similar repositories for book-opensips-101
Users that are interested in book-opensips-101 are comparing it to the libraries listed below. We may earn a commission when you buy through links labeled 'Ad' on this page.
Sorting:
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆37Jun 7, 2018Updated 7 years ago
- sample configurations and settings for opensips for various use-cases☆67Sep 10, 2019Updated 6 years ago
- ☆26Jun 5, 2013Updated 12 years ago
- Kamailio config for public/private proxy with rtpengine☆15Nov 21, 2023Updated 2 years ago
- SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality☆21Jul 5, 2024Updated last year
- GPU virtual machines on DigitalOcean Gradient AI • AdGet to production fast with high-performance AMD and NVIDIA GPUs you can spin up in seconds. The definition of operational simplicity.
- Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆14Apr 24, 2015Updated 11 years ago
- React SIP user agent☆56Oct 21, 2019Updated 6 years ago
- Multiparty-meeting (mediasoup) SIP gateway using Kurento☆28Apr 4, 2019Updated 7 years ago
- Connect as WebRTC with FreeSWITCH using SIPjs☆11Apr 9, 2019Updated 7 years ago
- Fork of the Asterisk VOIP software with support for the Opus codec☆14Sep 11, 2012Updated 13 years ago
- Testprogram for libre and librem☆11Oct 20, 2021Updated 4 years ago
- Load-balancing SIP proxy for Freeswitch☆29Jun 28, 2016Updated 9 years ago
- SIPp scenarios I use for testing SIP stuff☆305Jul 16, 2024Updated last year
- nodejs client for accessing rtpengine via ng protocol☆24Jul 18, 2023Updated 2 years ago
- Wordpress hosting with auto-scaling - Free Trial Offer • AdFully Managed hosting for WordPress and WooCommerce businesses that need reliable, auto-scalable performance. Cloudways SafeUpdates now available.
- FreeSWITCH dialer program for VoIP performance tests☆48Sep 15, 2017Updated 8 years ago
- CUCM Bulk TFTP Upload☆12Jun 4, 2018Updated 7 years ago
- Continously watches and notifies if new MACs are discovered in the local LAN.☆15Dec 17, 2012Updated 13 years ago
- WebSocket Command Line Tool For SIP And Template Data☆44May 23, 2024Updated last year
- C++ network framework，currently we support http, https, websocket☆13Sep 25, 2019Updated 6 years ago
- Bare UI for Push-To-Talk and SIP Call/Messaging☆23Nov 25, 2019Updated 6 years ago
- Telnyx JavaScript library for building WebRTC apps with SIP.js☆15Apr 2, 2026Updated last month
- Send xConfiguraiton & xCommand to Cisco TelePresence Endpoints i.e. DX70 & DX80☆13Jun 30, 2016Updated 9 years ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆44Aug 19, 2014Updated 11 years ago
- GPUs on demand by Runpod - Special Offer Available • AdRun AI, ML, and HPC workloads on powerful cloud GPUs—without limits or wasted spend. Deploy GPUs in under a minute and pay by the second.
- This is an Ansible module which connects to the target CUCM servers via SSH in order to execute CLI commands and gather outputs☆12Jul 30, 2021Updated 4 years ago
- A WebRTC-SIP gateway for AVM Fritz!Box based on Kamailio and rtpengine.☆23Dec 1, 2025Updated 5 months ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Jul 30, 2022Updated 3 years ago
- Integra Workflow Designer for Asterisk☆14Jun 7, 2022Updated 3 years ago
- Chef cookbook to install FreeSWITCH☆25Dec 22, 2015Updated 10 years ago
- Make test calls to our PSTN phone numbers and SIP/VOIP addresses☆16Mar 1, 2023Updated 3 years ago
- Web dialer SIP based on JsSIP☆17Mar 24, 2014Updated 12 years ago
- Uses BeautifulSoup and RegEx to pull and parse data directly form Cisco IP Phones☆16Jun 2, 2021Updated 4 years ago
- Tool for native number resolution - Cisco Unified Communications Manager - CURRI☆16Aug 11, 2018Updated 7 years ago
- 1-Click AI Models by DigitalOcean Gradient • AdDeploy popular AI models on DigitalOcean Gradient GPU virtual machines with just a single click. Zero configuration with optimized deployments.
- An OpenAPI specification for the Vimeo API.☆16Dec 13, 2023Updated 2 years ago
- A real time stream trans to other application or server for FreeSWITCH☆42Feb 8, 2025Updated last year
- Demo example applications using libre☆16Jan 27, 2021Updated 5 years ago
- Simple Click2Call for Asterisk PBX platform☆14Jan 9, 2015Updated 11 years ago
- A Fork from http://git.freeswitch.org/git/freeswitch-contrib/tree/dvarnes/java/esl-client☆244Apr 29, 2022Updated 4 years ago
- The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.☆473Apr 19, 2026Updated 2 weeks ago
- SIP to SMPP Gateway☆16Jun 17, 2015Updated 10 years ago