Asterisk 22 LTS: New Features and Upgrade Guide
Asterisk 18 reached end of life in October 2025. It no longer receives patches of any kind, including security fixes. If you are still running it, migrating to a supported branch is overdue.
Asterisk 22 is the current Long Term Support release, out since October 2024, and the recommended target for most deployments.
Version landscape in 2026
| Version | Type | Status | End of Life |
|---|---|---|---|
| 18.x | LTS | End of life | Oct 2025 |
| 20.x | LTS | Fully supported | Oct 2027 |
| 21.x | Standard | Security fixes only | Oct 2026 |
| 22.x | LTS | Fully supported | Oct 2029 |
| 23.x | Standard | Fully supported | Oct 2027 |
Asterisk 22.x is the safe long-term choice. Full support runs through October 2028, with security-only maintenance continuing until October 2029.
If you are on Asterisk 21, it transitions to end of life in October 2026 — less than five months away.
What is new in Asterisk 22
ARI tone detection
Setting TONE_DETECT on a channel in an ARI application now causes a
ChannelToneDetected event to fire when the tone is heard. Previously, detecting
in-band tones from an ARI context required workarounds. The new event integrates
cleanly with event-driven call flows and is useful for IVR systems, fax detection,
and any application reacting to call-progress signals.
Recording duration in RECORD_TIME
The Record() dialplan application now sets RECORD_TIME to the duration of
the completed recording in milliseconds. If you feed call metadata into a database
or analytics pipeline, you no longer need to open the audio file to measure its
length. The variable is available immediately after Record() returns.
STIR/SHAKEN hardening
Attestation level propagation had several edge-case bugs that could silently produce incorrect signed identity headers. Version 22 fixes the propagation logic and adds config-validation checks that catch malformed STIR/SHAKEN configuration files at startup rather than crashing mid-call.
The companion Asterisk 23 release (October 2025) added the
STIR_SHAKEN_ATTESTATION dialplan function, which lets you suppress or override
attestation on a per-call basis without modifying the endpoint profile. If you are
planning a fresh install, 23 is worth evaluating for this alone.
PJSIP codec negotiation fix
Endpoints advertising 48 kHz audio while only supporting 8 kHz DTMF payloads could fail to establish correct tone transmission. Version 22 resolves the negotiation mismatch. This was a difficult bug to diagnose because calls connected and audio worked, but DTMF was unreliable under specific codec combinations.
SIP URI crash fix
Malformed Contact or Record-Route URIs in incoming SIP requests could crash
the Asterisk process when res_resolver_unbound was loaded. The fix adds null
pointer checks and graceful error handling at the resolver boundary. Deployments
exposed to untrusted SIP traffic should treat this as a priority update.
Upgrade path
There is no in-place upgrade between major Asterisk versions. The standard approach is a fresh install followed by configuration migration.
From Asterisk 18: Migrate directly to 22.x. Review the breaking changes in the
Asterisk 22 documentation before touching pjsip.conf, extensions.conf, or any
custom AGI/ARI integrations. The PJSIP configuration model has evolved since 18.
From Asterisk 20: No urgent pressure — 20 remains fully supported through October 2026, with security maintenance until October 2027. Schedule a move to 22 while 20 is still active so you have a safe fallback window.
From Asterisk 21: Migrate before October 2026. You have a few months.
The Asterisk version table and download archive are the authoritative references for EOL dates and release notes.
Running a supported Asterisk branch is not optional for any internet-facing deployment. Version 18 is no longer safe. Asterisk 22 LTS is the current standard, and planning the migration now — before you are forced into it by a security incident — is the right call.