Asterisk PBX

AstriCon 2026 is coming to Pasadena, CA on March 5–6, presented alongside SCaLE. We really shouldn’t do this, but we just cracked open the agenda files and had to give you a sneak peek 🤫 Session Preview: Diego Gosmar, Chief AI Officer at Voice Interoperability is here to shake up the AI game. He’ll dive into how to create a collaborative multi-agent AI conference using Asterisk, where multiple AI agents team up to chat in a shared environment. With the Open Floor Protocol, he’ll show you how to move from boring, isolated exchanges to a dynamic, coordinated, and seriously collaborative AI experience. If you haven’t registered yet, there’s still time to secure your spot: https://hubs.ly/Q03Sgk8v0

If you’ve worked with FreePBX or Switchvox, you know how simple it is to set up SIP trunks using a token or keycode. That same process is now available in Asterisk with the res_pjsip_config_sangoma module, which lets you quickly connect SIPStation or VoIP Innovations trunks without extra manual setup. Note: this module supports PJSIP only, not chan_sip. Check out the full setup guide on our blog: https://hubs.ly/Q03QNWB20

Asterisk 18 is now End of Life. Asterisk 21 moves to Security Fix Only. Asterisk 18 has officially reached End of Life, which means no more updates, bug fixes, or security patches. It’s time to upgrade to a supported version to keep your system secure and running smoothly. Asterisk 21 has also entered Security Fix Only status. It will still receive critical security patches until October 2026, but no new features or non-security updates. We encourage all users to upgrade to a currently supported version of Asterisk. Learn more: https://hubs.ly/Q03Py9nz0

Still thinking about it? The AstriCon 2026 Call for Speakers is still open. Apply for a full session (45 minutes) or a lightning talk (15 minutes) and join us March 5–6 in Pasadena, CA, alongside SCaLE. Apply to speak at AstriCon 2026: https://hubs.ly/Q03Nlz1G0 To better align with the main SCaLE event we are extending our submission deadline to December 1st, but don't wait until then to submit your speaking proposal! The sooner you submit, the better!

Don’t overthink your SIP trunking setup. Join us to learn how to simplify Asterisk configuration with Sangoma’s new Trunking Module and token-based setup. We’ll show you how to make setup faster, easier, and more reliable, even if you’re new to SIP trunking. Get the confidence to set it up right the first time. Save your spot: https://hubs.ly/Q03Mxwx30

Setting up SIP trunking with Asterisk doesn’t have to be complicated. Join our upcoming webinar to learn how to connect Sangoma SIP trunking with Asterisk in just a few simple steps. We’ll walk through the new Sangoma Trunking Module, show you how token-based setup works, and help you take the guesswork out of configuration. Register now: https://hubs.ly/Q03MxpkV0

The AstriCon 2026 Call for Speakers is open! We’re looking for talks on AI, Security, E911, STIR/SHAKEN, Scaling, FreePBX, and more. Sessions can be full length (45 minutes) or lightning (15 minutes). ⚡ Join us March 5–6 in Pasadena, CA, alongside SCaLE. Submit your talk by Nov 1 and share your story with the open source communications community. Apply to speak now: https://hubs.ly/Q03Lr7Pj0

Good news: exchanging media between Asterisk and your apps just got easier! The new chan_websocket module lets you move audio over WebSockets without dealing with RTP headaches. It works with popular codecs, supports secure connections, and handles timing and silence for you. You can even send files or AI responses and Asterisk will play them back cleanly. Control calls with commands like ANSWER, HANGUP, or PAUSE_MEDIA and get real-time events back on the same path. Incoming or outgoing connections are supported, making integration more flexible. More info is available on the Asterisk documentation page and code samples on GitHub. ➡️ https://hubs.ly/Q03J003z0

The AsteriskVoiceBridge is here—an open-source Go app that brings together Asterisk, Deepgram, and OpenAI to create real-time voice AI. How it works: - A call comes in and audio is transcribed by Deepgram. - The text goes to an AI agent (OpenAI), which figures out the best response or action. - The reply is turned back into speech by Deepgram and sent to the caller through Asterisk. - The system can even pick tools or next steps automatically, guided by a Voice Business Logic Controller. There’s a built-in demo IVR with options for sales, support, or accounts—showing how AI can handle real conversations and route calls on the fly. While it’s designed for learning (not production), it’s a powerful look at what’s possible with voice + AI today. 👉 https://hubs.ly/Q03GNCs30

Snooping and whispering with Asterisk ARI is possible using node-ari-client. Set up a service bridge, attach a snoop channel to monitor audio, and use an externalMedia channel to stream audio out. This lets your app listen, analyze, or respond. The original caller stays outside the service bridge so you can route them elsewhere, like into a conference or another app. To whisper back, make sure the caller’s channel is already receiving audio (e.g. bridged with another channel or in playback). More details here: https://hubs.ly/Q03xwS_M0 #Asterisk #VoIP #TelecomDev