AiPBX

PBX fine tuning & maintenance

Asterisk core tuning, SIP protocol timers, adaptive jitter buffering, night maintenance restart windows, log diagnostics, and spoken status announcements.

• Core tuning & diagnostics: PBX settings → Tuning & Ops (tab in /asterisk-settings).
• Spoken status messages: PBX settings → Ringtone (tab in /asterisk-settings).

Adaptive jitter buffer

Jitter buffering smooths out network packet arrival timing variations on VoIP calls.

Setting	Default	Range / Options	Notes
Jitter Buffer Mode	Adaptive (Active)	Active / Disabled (Off)	When disabled, audio packets pass without buffering
Maximum Buffer Size	200 ms	50–2000 ms	Upper ceiling for buffered audio frames
Initial Buffer Size	100 ms	20–1000 ms	Starting buffer depth when a call establishes

• Fax exemption: Jitter buffering is automatically bypassed for faxes. Reshaping audio packet timing disrupts T.30 / T.38 modem handshakes and causes faxes to drop.

SIP protocol timers & Max-Forwards

Fine-tune SIP signaling behavior for high-latency trunks or deep routing chains:

Setting	Default	Range	Description
Max-Forwards Limit	70	1–255	Maximum SIP proxy hops allowed before a request is rejected; prevents infinite forwarding loops
Timer T1	500 ms	50–10000 ms	Base round-trip time (RTT) estimate for SIP transactions; increase for high-latency links (satellite, cellular, or VPN trunks)
Timer B	32000 ms	500–64000 ms	INVITE transaction timeout (standard 64 × T1) before failing an unacknowledged invite

Changes to Timer T1 and Timer B are written to PJSIP system configuration and take effect after the next Asterisk restart.

Live diagnostics & logs

10-minute SIP debug

Click Enable SIP Debug (10 Minutes) to turn on verbose PJSIP packet dumping in Asterisk.

• Displays a live countdown badge showing remaining minutes.
• Writes every SIP message sent and received to the Asterisk log (pjsip set logger on).
• Automatically switches itself off after 10 minutes to protect disk space and CPU performance.
• Can be stopped at any time with Stop SIP Debug.

On-demand log rotation

Click Rotate Asterisk Logs Now to rotate all Asterisk log files at once (logger rotate), for example before reproducing a problem. Calls are not affected.

Safe night maintenance restart window

Ubuntu automatic security updates (unattended-upgrades) periodically install updated system libraries (such as OpenSSL, glibc, or SQLite). Since version 1.50.7, needrestart leaves Asterisk alone to prevent interrupting calls during the day.

The Night Maintenance Restart Window lets Asterisk safely restart to load updated libraries during quiet hours.

Setting	Default	Description
Night Restart Status	Passive (Off)	Enable or disable safe night restarts
Window Start	03:00	Start of the maintenance window in server local time
Window End	05:00	End of the maintenance window in server local time

The window can cross midnight (e.g. 23:00 to 04:00).

Execution conditions

The task dispatcher evaluates the restart window every minute. A restart is triggered only when all of the following conditions are met:

1. Feature is enabled: Night restart is switched on.
2. Current time is in window: Local server time is between the configured start and end times.
3. At most once a day: A restart has not already occurred today.
4. A security update is waiting: Asterisk still uses a security-relevant library that an update replaced (OpenSSL, GnuTLS, the C library, SRTP, curl, libxml2 …). Updates of other libraries (libsystemd, libudev …) change nothing for calls and never cause a restart.
5. Zero active calls: The active call count on Asterisk is strictly 0.
6. Call count is verifiable: If Asterisk cannot be queried or the call count cannot be determined, the restart is aborted.

Every automated night restart is recorded in the system audit log.

Failed registration alarm

Under PBX settings → Tuning & Ops, the Failed Registrations Threshold (per 15 min) setting (default 10, range 1–1000) controls the security threshold for SIP login attacks. When failed registration and authentication attempts in /var/log/asterisk/security.log exceed this number within 15 minutes, the Health check Failed registrations triggers an alarm in the alert bell and e-mails administrators.

Spoken status instead of tones

PBX settings → Ringtone (applied with Apply).

When an internal call to an extension ends because the extension is busy, does not answer, or is unreachable (and no voicemail or call forwarding takes the call), callers can hear a spoken explanation in their own language instead of only a busy or congestion tone:

Status	Message text
Busy	“The person at extension 3001 is on the phone. Please try again.”
No answer	“The number you are calling is not answering. Please try again.”
Unreachable	“The person at extension 3001 is unavailable. Please try again.”

• Billing safety: For calls originating from external trunks, the spoken explanation is played as early media before the call is answered. The caller hears the explanation, but the trunk call is never answered, preventing call charges.
• Default: Off (Tone only) for all three conditions. Each can be set independently to Tone only or Spoken message.

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