If you’ve ever needed help with Asterisk, you should sign up for this. On January 20 at 12:00 PM Eastern, our Asterisk expert, Josh, will walk through all the ways you can get support. Register now and share with anyone you know who runs Asterisk. https://t.co/NiBvMaory0

OUT NOW: The 2026 AstriCon Agenda Co-located with SCALE, join us in Pasadena, CA on March 5–6 for two full days of enlightening sessions. From scaling telephony platforms to real-time Voice AI. 👉 Check out the full agenda here: https://t.co/j3tXpIvrl0

AstriCon 2026 is getting close; here’s another agenda peek. Noah King from Horizon3 will break down real FreePBX vulnerabilities and how to find, validate, and handle issues like SQL injection, file upload flaws, and auth bypasses. Register: https://t.co/WKI67CnZjE

Still peeking inside the agenda files of AstriCon 2026... Alex Balashov of Evariste Systems will unpack the shift in the American PSTN core from SS7/TDM to modern IP peering—what changed, why it matters, and how it shapes today’s networks. https://t.co/LFTc3bp21D

We built a new Taskpool API for Asterisk to replace heavy threadpool handling for short tasks — and the performance jump is huge: • 2–3× more tasks/sec • Up to 20× gains on some systems • Early users seeing 20–30% lower CPU More here: https://t.co/yuc1X2j6cl

If you haven’t registered yet, there’s still time to secure your spot:

AstriCon 2026 hits Pasadena March 5–6 with SCaLE. We really shouldn’t, but we cracked open the agenda files for a sneak peek 🤫 Diego Gosmar of Voice Interoperability will show how Asterisk + OFP power collaborative multi-agent AI in a shared conversational space.

Running Asterisk? You can auto-configure SIP trunks for SIPStation or VoIP Innovations using the res_pjsip_config_sangoma module. Quick setup, no manual edits, PJSIP only. Check out the full setup guide on our blog: https://t.co/ClwuhC9fRf

Asterisk 18 has officially reached End of Life, and Asterisk 21 has entered Security Fix Only status. Upgrade to a currently supported version of Asterisk to keep your system secure and running smoothly. Learn more: https://t.co/gx5gTeh8Fg

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. Deadline to speak is now Dec 1. Apply now: https://t.co/ZPkQtvKgl4

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. https://t.co/dp5TCZrb0m

Join our upcoming webinar to learn how to connect Sangoma SIP trunking with Asterisk in 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. https://t.co/9eLBL3TQ1R

AstriCon 2026: Call for speakers is open! Got insights on AI, Security, FreePBX, or scaling communications? Submit your talk—full (45 min) or lightning (15 min)—by Nov 1. Apply now: https://t.co/9mL3nl78dD

Exchanging media between Asterisk and apps is easier with chan_websocket. Send audio over WebSockets, skip RTP hassle, use popular codecs, control calls, and get events back in real time. ▶️Check it out: https://t.co/KQDEEVrdpd

The AsteriskVoiceBridge is an open-source app that links Asterisk, Deepgram, and OpenAI for real-time voice AI. Call in → AI processes → Voice replies out. Includes a demo IVR with sales, support, and accounts. Note: It’s for learning, not production. https://t.co/ZLAZyAyx3k

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. More details here: https://t.co/qNoPssXU54

New in Asterisk 20.14.0, 21.9.0, and 22.4.0: You can now send ARI REST requests directly over the same WebSocket you already use to recieve events. No more juggling HTTP and WebSocket connections. Learn more: https://t.co/NCYAXEba3M

If you're new to audio over websocket and come from the #telephony world, you might assume that this works similarly to an RTP stream where the audio is framed and timed for real-time listening. This isn’t always the case. Read more at https://t.co/WUz3ZZPuGf #Asterisk

We are moving exclusively to Fosstodon. If you'd like to follow us for news and updates, you can find us here: https://t.co/up0Eq7LDGh #social

Asterisk now supports the SHA-256 and SHA-512-256 SIP authentication digest algorithms. Read more at