QUICK SUMMARY
Asterisk 23.0.0 trims outdated components, fixes long-standing issues, and improves system reliability. Here’s the simple rundown of what’s changed, why it matters, and how to get it running properly.
Asterisk just dropped version 23.0.0, and before you ask, no, Asterisk 23 isn’t another ‘
’ pretending to be innovation. This release moves beyond what Asterisk 22 started, targeting the performance gaps and reliability issues that show up when systems are under real load.
Let’s see
❓ What’s new,
✅ What’s improved, and
⚠️ Why teams running serious VoIP workloads should pay attention.
New Features in Asterisk 23
From the official release notes:
1. New global option: log_unpause_on_reason_change in app_queue
A new global setting in the app_queue.c module: when enabled, every time a pause in the queue is lifted with a different “reason” for the pause, an UNPAUSE event is logged.
Why it matters: Enables finer tracking/monitoring of queue behaviour; e.g., when agents pause/resume for different reasons, distinct events help analytics and audit trails.
2. Custom tone for WaitExten (dialplan builtin)
Under pbx_builtins, you can now override the tone used while waiting for digits in WaitExten by specifying an argument for the d option.
Why it matters: Flexibility for IVR/auto-attendant setups, you might want non-default tones or signals while waiting for user input, to improve UX or align with a brand/admin tone.
3. Improved tone detection: auto-stop option in res_tonedetect
The TONE_DETECT function/module got an e option: when specified, detection will automatically stop once the desired number of matches is reached, avoiding the need for manual disable.
Why it matters: Prevents race conditions in dialplan logic; reduces the risk of lingering tone-detection modules unnecessarily tying up resources.
4. Sorcery backend improvements: prevent duplicates / auto-create missing objects
For users using multiple writable backends (e.g., ASTDB plus realtime), a new option update_or_create_on_update_miss = yes in sorcery.conf allows automatically recreating missing objects when updates fail.
Why it matters: Improves stability for systems using real-time datastore backends; avoids data inconsistencies or broken objects due to backend failure/temporary outage.
5. WebSocket channel driver (chan_websocket) additional URI parameter support
A new option v on the Dial application allows specifying additional URI parameters for outgoing WebSocket connections.
Why it matters: Enhances flexibility for Web-RTC or browser-based softphone integrations; you may need to pass extra parameters (e.g., “room id”, “device id”, metadata) when connecting.
6. app_chanspy / ExtenSpy: option to not auto-answer channel
The ‘N’ option is now available: when used, ChanSpy will not automatically answer the monitored channel.
Why it matters: In monitoring/quality assurance use-cases, you might prefer to silently tap into a live call without causing an audible ‘answer’ tone or being detected. This helps make monitoring less intrusive and more seamless.
What would Asterisk 23 look like in your setup with expert tuning? Ecosmob can show you.
Key Fixes and Improvements in Asterisk 23
Asterisk 23 isn’t about big shiny features but about fixing the things that quietly cause headaches in real deployments. From smarter bridging behavior to cleaner WebRTC handling, better logging, and fewer random crashes, this release focuses on making the platform more stable and predictable where it counts. Here’s a quick look at the key improvements.
Bridge Behavior
Fixed a bug where BRIDGE_NOANSWER was not always honored. In some cases, Asterisk would answer a channel even when instructed not to.
This is important for call flows where you want to bridge silently or control exactly when a channel is “answered.”
Frame Drop Handling
Improved func_frame_drop to better handle memory allocation failures, avoiding potential crashes or memory corruption.
Added debug messages that log dropped frames, so it’s easier to trace and diagnose why frames are being dropped.
This helps in high-load/real-time audio scenarios where frame drops can affect call quality; better logging means more visibility into problems.
WebSocket Channel Fixes
Fixed a buffer overrun bug in chan_websocket when processing TEXT frames, which could lead to heap corruption.
Corrected improper handling of the “opus” codec in the WebSocket channel driver.
Ensured that RTP isn’t sent before DTLS negotiation completes (important for secure media setup).
These improvements strengthen WebRTC development or browser-based connections, making them more stable and secure.
Analog Signaling (DAHDI / SIG-ANALOG)
Fixed a segmentation fault (crash) in sig_analog when strcmp was called on a NULL pointer.
Corrected behavior so that dial-mode settings in DAHDI (chan_dahdi) reset properly between calls. Previously, certain dial-mode changes would “stick” across calls.
If CallerID is disabled (usecallerid = no), the system no longer “spills” caller ID information.
These fixes make analog channel behavior more robust and predictable, especially in legacy setups.
Configuration / Template Handling
Fixed how deep/wide template inheritance is saved in config files (via ast_config) so that variable inheritance works correctly.
This addresses issues where configuration inheritance logic caused unintended or incorrect template behavior, improving reliability when using templates for config management.
Sorcery Backend Reliability
Prevents duplicate objects when using multiple writable backends (e.g., ASTDB + realtime), avoiding inconsistent data states.
Adds option update_or_create_on_update_miss = yes in sorcery.conf. When enabled, if an update fails because the object is missing in a backend, Asterisk will try to create it.
This is very helpful for systems relying on multi-backend persistence, making them more resilient to temporary backend failures.
Tonedetect / DSP Logging
Improved debug logging in tone detection (dsp.c) to be more useful: it suppresses redundant logs, shows hit count, and uses cleaner logging levels.
This means better observability for tone-detection use cases, such as IVR systems or custom dialplans.
ARI Device State Cleanup
Fixed a bug in the res_stasis_device_state module where ARI device states weren’t being deleted from cache after a restart, leading to stale or incorrect state.
This cleanup ensures that ARI-based applications that track device states don’t get inconsistent or “zombie” state.
Voicemail Module Cleanup
Removed deprecated options in app_voicemail (like maxmessage and minmessage).
This helps simplify the configuration and reduces legacy clutter.
CLI / Logger Cleanups
Removed redundant or deprecated CLI commands (like old “no debug channel”).
Cleaned up deprecated logging config options (e.g., rotatetimestamp).
Tidied up deprecated dialplan functions in func_dialplan.
These improvements streamline the core CLI and configuration, making Asterisk leaner and more maintainable.
AMI / ARI Crash Fixes
Fixed a segmentation fault (SEGV) in res_pjsip_authenticator_digest when get_authorization_hdr() returns NULL.
Improved ARI: added a new command (REST endpoint) to signal “progress” to a channel.
These fixes improve the stability of both control-plane (ARI) operations and SIP authentication workflows.
Not sure if your current configs fit well with Asterisk 23’s changes?
Asterisk 22 vs Asterisk 23.0.0 Comparison
Asterisk 22 was all about delivering fresh, useful features, and we explored those in depth in our Asterisk 22 guide. Asterisk 23.0.0, however, shifts gears. This release prioritizes platform refinement: fixing recurring challenges, improving stability, and ensuring the system performs more reliably in production environments.
Here’s a quick snapshot of how both versions differ where it counts.
| Aspects | Asterisk 22 | Asterisk 23.0.0 |
| Release Focus | Feature additions | Stability + fixes |
| Tone Detection | Basic detection + ARI event | Controlled detection with new “e” option |
| Queue / Agents | Standard behavior | Better upause logging |
| Websockets / WebRTC | Minor updates | URI, passthough, DTMF handling fixes |
| Security / RTP | URI vuln fix | No RTP pre-DTLS |
| Config / Sorcery | Standard config logic | Cleaner inheritance & no duplicate objects |
| Stability / Crashes | Single major fix | Multiple crash + memory stability fixes |
| Deprecated Items | – | ‘users.conf’ removed |
complicated, nothing dramatic. You just set up your system, pull the new source, build it, and make sure everything
Install Asterisk 23 Version
If you’ve installed Asterisk before, you already know the drill, but Asterisk 23.0.0 brings just enough cleanup and changes that it’s worth a fresh look. Nothing fires up the way it should. Here’s a quick sneak peek at what you’ll need before installing Asterisk 23.
Asterisk 23.0.0 Installation Prerequisites
Before installing Asterisk 23, make sure your server has the essential development tools and libraries needed for a stable build.
You’ll need:
- A Linux system (Ubuntu, Debian, CentOS, Rocky, Alma)
- Development tools (GCC, Make, Build Essentials)
- libjansson-dev (JSON support)
- libxml2-dev (XML parsing)
- SQLite3 development headers
- OpenSSL development libraries
- PJSIP libraries (or use the bundled build)
- ncurses and libedit for CLI handling
Asterisk 23 Step-by-Step Installation
Follow these steps to install Asterisk 23.0.0 cleanly from source:
1. Download Asterisk Source
2. Install Additional Prerequisites
Fetch MP3 support and run the helper script:
3. Configure Build
Optional module selection:
4. Compile & Install
5. Start the Asterisk Service
6. Verify Installation
If you land in the Asterisk CLI and see version 23.0.0, the installation is complete.
Should You Upgrade to Asterisk 23?
Upgrading isn’t just about chasing the newest version number. It’s about whether the update actually solves the problems you’re dealing with, or if it introduces more work than it’s worth. Asterisk 23 is very much a “clean-up and tighten things up” release, not a flashy feature showcase. So the real question is: does your environment benefit from those fixes and improvements right now?
Here’s the straight answer based on common scenarios:
Upgrade Immediately If You’re Facing Any of This:
- WebRTC or browser-based calling is a major part of your setup
(Asterisk 23 has important WebSocket + DTLS behavior fixes.) - You’re using tone detection heavily in IVRs or custom dial plans
(The new control options reduce false triggers and race issues.) - You rely on queues and agent state tracking
(More detailed unpause events = better analytics + fewer blind spots.) - Your system has experienced random crashes or memory leaks
(Asterisk 23 fixes several subtle but real-world crash conditions.)
Upgrade When It’s Convenient If:
- Your Asterisk deployment is stable and not showing warning signs
- Your team wants cleaner config handling and better Sorcery behavior
- You prefer staying closer to the latest branch for long-term planning
- You’re preparing for new features or integrations later in the year
Asterisk 23 isn’t urgent unless you need the improvements, but it is a noticeably cleaner engine under the hood.
Consider Waiting If:
- You’re running a mission-critical system with zero wiggle room for downtime
- Your setup includes heavy custom modules that need testing before any version jump
- You rely on deprecated components removed in Asterisk 23 (like users.conf)
- Your environment needs LTS guarantees rather than standard releases
For these setups, a slower controlled upgrade path is the smarter move.
Expecting better stability from Asterisk 23? See what this upgrade can do to your system.
Asterisk 23 Migration Best Practices
Upgrading to Asterisk 23 isn’t complicated, but doing it the smart way saves you from those “why did we rush this?” moments.
Here’s how to move to 23 without breaking anything you actually care about.
Before You Upgrade
Test it first, seriously.
Spin up a dev or staging environment and run your usual call flows. Asterisk 23 removes a few old pieces and tightens behavior in others, so catching quirks early is better than discovering them mid-shift.
Backup everything.
Config files, custom modules, AGI scripts, voicemail, trunks, if it matters, back it up.
You don’t want to play “hunt the missing setting” after the upgrade.
Document your setup.
List out modules, custom patches, endpoints, codecs, and any odd configurations your team has added over the years. It’ll help you trace issues quickly if something behaves differently.
Review deprecated components.
Asterisk 23 removes items like users.conf and older dialplan elements. Clear them out now instead of letting them bite you post-upgrade.
During the Upgrade
Follow the standard upgrade steps.
Don’t improvise unless you absolutely know what you’re doing. Asterisk upgrades reward consistency, not creativity.
Watch your logs.
Asterisk 23 is more vocal about errors and deprecated behaviors. If something throws warnings during startup, fix it now, not in production.
Validate your critical call flows immediately.
Inbound, outbound, IVR, queues, WebRTC, and ARI apps run through every high-traffic path.
Check all integrations.
CRM, billing system, monitoring tools, recording servers, confirm everything is still talking to Asterisk the way it should.
After the Upgrade
Run full call simulations.
Test different scenarios: high load, mixed codecs, transfers, queue churn, tone detection, etc. Make sure everything feels stable.
Monitor performance for at least 24–48 hours.
Check CPU, memory, RTP flow, logs, PJSIP events, and queue behavior. Asterisk 23 has several under-the-hood fixes; if something is off, you’ll spot it here.
Look for new warnings.
Asterisk 23 cleans up older config logic, so some setups will show warnings you’ve never seen before. Don’t ignore them.
Confirm all deprecated items are fully removed.
If you still have leftover configs referencing outdated modules or functions, clean them up before they create long-term confusion.
Optimizing Asterisk 23 Performance
Asterisk 23 already cleans up a lot under the hood, but the real performance gains come from how you configure and tune it. If you want smoother call handling, fewer random hiccups, and more predictable behavior under load, these are the areas worth tightening up.
1. Get Your PJSIP Setup in Order
Asterisk 23 includes important fixes in WebSocket, DTLS, and SIP handshake behavior, but it won’t magically fix bad configs.
Focus on:
- Reviewing mixed codec scenarios (Opus + narrowband DTMF can still be messy if misconfigured)
- Making sure transport reuse is enabled where appropriate
- Cleaning up unused endpoints and templates
- Ensuring your DTLS settings are correct if you use WebRTC
Why this matters:
PJSIP is still one of the highest-impact components. Clean configs = fewer retries, fewer timeouts, and much more stable media handling.
2. Tone Detection: Use the New Controls
Asterisk 23 adds better control over tone detection with the new e option.
Use it when:
- You want tone detection to stop after a certain number of hits
- You need cleaner IVR flows
- You want to avoid tone-detection processes running longer than necessary
Translation:
Less CPU churn. More accurate flow handling. Fewer dialplan surprises.
3. Keep Your Logging Tight and Useful
Asterisk 23 has improved logging around frame drops, DSP hits, and queue unpause reasons.
Use them.
Best practices:
- Enable a dedicated debug log (not everything in the main log)
- Rotate logs aggressively to avoid I/O slowdowns
- Monitor WebRTC or PJSIP logs during peak hours to catch codec mismatches early
Why:
Good logs prevent 90% of troubleshooting nightmares, especially on WebRTC-heavy deployments.
4. Check Your Sorcery Backends
Asterisk 23 added logic to avoid duplicate/missing objects in multi-backend setups.
Do this:
- If you’re using multiple writable backends (Realtime + ASTDB), enable:
update_or_create_on_update_miss = yes - Clean up stale backend objects
- Validate that template inheritance isn’t pulling junk from old configs
Result:
Fewer “ghost” endpoints, cleaner reloads, and more reliable runtime behavior.
5. Trim the Fat (Modules You Don’t Need)
Asterisk 23 removed some deprecated components, but you should go further.
- Disable modules you don’t use (e.g., CEL backends, legacy codecs, old apps)
- Keep only PJSIP, ARI, and essential channel drivers
- Re-run menuselect occasionally as new modules are added or removed
Why:
The leaner Asterisk gets, the better it runs, especially on busy SIP servers.
6. Watch RTP and DTLS Behavior Closely on WebRTC
Asterisk 23 fixed the issue where RTP could leak out before DTLS finished negotiating.
Still, check:
- DTLS certificates
- ICE/STUN/TURN settings
- Browser-to-server negotiation consistency
- Codec order (Opus → PCMU/PCMA fallback)
WebRTC is unforgiving.
A small config mistake = jitter and latency, silent audio, or failed handshakes.
7. Load Testing Isn’t Optional Anymore
With all the internal improvements in Asterisk 23, you’ll get the most out of the upgrade if you test your environment with real traffic patterns.
Run tests for:
- High concurrent call volume
- Rapid agent logins/logouts
- Tone-heavy IVRs
- Transfers and call parking
- Long-running calls
Small issues often show up only when the system is busy, not during casual testing.
Asterisk 23.0.0 isn’t the kind of release that makes noise; it’s the kind that makes systems run better. For teams already running Asterisk 22, this release further tightens reliability with cleaner internals and smarter fixes where they matter. If your communication stack needs stability, smoother performance, and fewer late-night surprises, this release is worth serious consideration. And if you’re aiming for a trouble-free upgrade with the best performance you can squeeze out of Asterisk, now is the right time to move with the right guidance behind you.
If you’re planning to move to Asterisk 23 and want the transition to be smooth, stable, and fully optimized for your environment, reach out for expert guidance. We’ve got you covered! Get support!
FAQs
What’s the biggest change in Asterisk 23.0.0 compared to previous versions?
Asterisk 23 isn’t a feature-heavy release; it focuses more on stability, cleaned-up modules, better tone detection control, improved WebSocket/WebRTC handling, and multiple crash/security fixes.
Is Asterisk 23.0.0 an LTS release?
No. Asterisk 23 is a Standard release. If you need long-term support, you should stick with the current LTS branch unless the new fixes in 23 directly benefit your environment.
Does Asterisk 23 improve WebRTC performance?
Yes. It includes several important fixes for WebSocket behavior, DTLS negotiation, and codec handling, making WebRTC setups more stable than in earlier releases.
Will upgrading to Asterisk 23 break my existing dial plans?
Not usually, but Asterisk 23 does remove some deprecated components like users.conf and old dialplan functions. If you're using legacy elements, test your configs before upgrading.
Is the installation process different for Asterisk 23?
The installation steps are almost identical to previous versions. The main difference is that some older modules aren’t available anymore, which makes the build slightly cleaner.
