PBX Settings

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Phone Settings

Microphone The change takes effect on the next call.

Speaker

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Ring Volume

PBX Settings

PJSIP & Global Communication Settings

PBX Configuration Guide

Configures general PBX options, SIP trunk IP/port definitions, WebRTC WSS ports, and ringtone/codec preferences. Branding & logo settings are now on a separate page (Administration → Branding & Appearance).

Ports for UDP, TCP and TLS are now on the Transports page: Transports

TLS Protocol Method Allowed TLS version negotiated with endpoints.

TLS SIP Domain (FQDN) Domain/FQDN for TLS transport (e.g. sip-tls.example.com). Empty leaves it default.

Audio Codecs

Opus aLaw (PCMA) uLaw (PCMU) G.722

For WebRTC (browser) softphones.

Desk Phone / Classic SIP Codecs

Opus aLaw (PCMA) uLaw (PCMU) G.729 G.722

For desktop/classic SIP phones (external to WebRTC).

SIP User Agent (Server Identity) The server name advertised in outgoing/incoming SIP messages (User-Agent / Server header). A plain value can be preferred to avoid disclosing version info.

Direct Media

NAT & RTP Symmetry

Extension Heartbeat Check (Qualify) ?How often, in seconds, Asterisk checks whether extensions are still connected (qualify_frequency). A lower value detects "offline" faster but generates more network traffic. Seconds (0 = disabled)

Internal Ring Timeout ?How many seconds a phone rings when one extension directly calls another (or a queue member). Seconds (extension-to-extension)

Outbound Trunk Ring Timeout ?How many seconds the far-end phone rings on outbound trunk calls. Seconds (outbound trunk)

Addresses the PBX uses

External IP Address The address phones and carriers outside see for this server, if it is behind NAT. It can be a host name, such as a dynamic DNS name.

Public media address (optional) Only if audio and video leave from a different address than SIP signaling. Empty uses the public address.

Local Network & VPN Subnets Subnet blocks

· Asterisk takes the change after Apply.

RTP (Media) Settings

Info

Controls which port range carries audio (RTP) and whether Asterisk rejects audio arriving from an unexpected source. This file (/etc/asterisk/rtp.conf) is now generated by the panel; manual edits on the server are lost on the next "Apply".

RTP Start Port ?Start of the port range reserved for audio streams. Every concurrent call consumes ports from this range.

RTP End Port ?End of the range. coturn (TURN relay) uses 13479-14999 — an overlap breaks WebRTC audio and is rejected on save.

Strict RTP (strictrtp) ?When enabled, Asterisk silently drops audio arriving from any source other than the learned remote address/port (attack protection). If the peer PBX sends media from an unexpected port this causes a silent "no audio" fault with no log trace — it can be turned off temporarily for diagnosis.

The Strict RTP setting takes effect with "Apply". A PORT RANGE change, however, requires a full Asterisk restart — the port pool is allocated once when the module loads.

T.38 UDPTL (Fax Media) Settings

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Defines UDP port ranges and forward error correction (FEC / Checksum) parameters for T.38 fax packets (UDPTL). This file (/etc/asterisk/udptl.conf) is dynamically managed.

UDPTL Start Port ?Starting UDP port for T.38 fax streams (default: 4100).

UDPTL End Port ?Ending UDP port for T.38 fax streams (default: 4999).

UDPTL Checksums

FEC Redundant Entries

FEC Span

Spoken status instead of tones

Info

When a call to an extension ends because it is busy, does not answer or cannot be reached (and no voicemail or forwarding takes it), the caller can hear a short message in the caller's language instead of only a tone. Calls from trunks hear it before they are answered, so they are not billed; faxes hear nothing. Takes effect after Apply.

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."

Softphone Ringtone & Dial Tone

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You can choose from sounds uploaded on the Announcements page for the incoming call ringtone and outbound dial tone in the header WebRTC softphone. Default internal sounds are used if left blank.

Incoming Call Ringtone

Outbound Dial Tone

Video Calling

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When video calls are on, the web phone gets a video call button next to the dial field and an incoming video call can be answered with video; the mobile apps offer Video call for colleagues. Video uses many times the bandwidth of audio; keep it off until the network is ready. Users also need the video phone mode.

Video Calling

Video codec One codec for all phones: Asterisk does not convert video, and a call whose two sides send different video codecs would be cut off. A phone without this codec talks without video. With H.264 the mobile app needs a phone that can encode H.264.

Maximum Resolution

Maximum Frame Rate

Language Settings

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The language used for Asterisk's own built-in voice prompts (queue hold announcements, "busy" tone, etc.). If the Inbound Route / IVR / Queue pages don't set a language, the default here is used. Important: Asterisk must be fully restarted for this setting to take effect (unlike other settings, "reload" alone is not enough).

System Default Language Installed sound packs are auto-detected from folders under /var/lib/asterisk/sounds/.

Time zone and tones

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One time zone for the server, the portal, call records and time conditions. Changing it restarts the database, Asterisk and the portal within a few seconds, and calls in progress drop, so choose a quiet moment. Working hours in time conditions then follow the new zone. The tones are the ring and busy sounds callers hear; changing them does not disturb calls.

Time zone

Ring and busy tones Choose your country, or the nearest one with the same tones.

If calls are in progress, end them for the time zone change

Answering Machine Detection (AMD)

Info

Global timing and word-detection parameters used by Asterisk's AMD() application during automated outbound campaigns (Auto Dialer). Times are in milliseconds (ms).

Initial Silence (ms) Max silence before greeting; if exceeded → MACHINE (default: 2500 ms)

Max Greeting Length (ms) Max length of initial greeting; if exceeded → MACHINE (default: 1500 ms)

After-Greeting Silence (ms) Silence after greeting required to classify as HUMAN (default: 800 ms)

Total Analysis Time (ms) Max total decision time before returning NOTSURE (default: 5000 ms)

Min Word Length (ms) Minimum continuous voice duration counted as a word (default: 100 ms)

Between-Words Silence (ms) Minimum silence gap separating two words (default: 50 ms)

Maximum Number of Words If greeting exceeds this word count → MACHINE (default: 2)

Silence Threshold (1–32767) Audio noise floor below which samples count as silence (default: 256)

Voicemail limits and PIN rules

Info

Limits of every mailbox and the rules a voicemail PIN must follow. A mailbox saved without a PIN gets a random one (it used to be the extension number, which anyone could guess); PINs changed from a phone must be as long as set here too. Takes effect after Apply.

1 mailbox(es) have a PIN that does not follow these rules (often the extension number). Set a new PIN on the extension or in My phone.

Longest message (seconds, 0 = no limit) 0–3600, default 0.

Shortest message kept (seconds) 0–60, default 3.

Messages per mailbox 10–9999, default 100.

Silence that ends a message (seconds, 0 = off) 0–60, default 2.

Wrong PINs before hanging up 1–10, default 3.

Shortest PIN (digits) 4–10, default 4.

PIN may not be the extension number

Signaling capture (SIP and call quality)

Info

Asterisk copies every SIP message and RTCP quality report to the aipbx-signaling service on this server, which stores them: the SIP exchange of each call and its packet loss and jitter (Reports → Call quality). Nothing leaves the server. Turning it on or off takes effect at once and does not touch calls in progress.

Capture 0 message(s) stored in the last 24 hours; newest: —.

Keep for (days) 1 to 3650. Older messages are deleted every few hours.

PBX Fine Tuning & Maintenance

PBX Fine Tuning Guide

Configure SIP protocol timers, adaptive jitter buffering parameters, night-time maintenance restart windows, failed-registration alarms, and live SIP diagnostics.

SIP Protocol Timers & Max-Forwards

Max-Forwards Limit Maximum number of SIP hops allowed (default 70). Prevents infinite forwarding loops.

Timer T1 (RTT Estimate, ms) Base SIP transaction timer (default 500 ms). Increase on high-latency satellite/VPN trunks.

Timer B (INVITE Transaction Timeout, ms) Timeout for unacknowledged INVITE transactions (default 32000 ms, 64*T1). Timer T1 and B take effect after the next Asterisk restart.

Adaptive Jitter Buffer

Jitter Buffer Mode Compensates for packet delay variation on VoIP calls. Automatically bypassed for faxes.

Maximum Buffer Size (ms) Upper threshold of audio frame buffering (default 200 ms).

Initial Buffer Size (ms) Starting buffer depth at call start (default 100 ms).

Safe Night Maintenance Restart

Night Restart Status Inside the window, Asterisk is restarted when a system update replaced a security-relevant library it still uses (OpenSSL, the C library, SRTP …), and only while no call is in progress (at most once a day). Otherwise nothing happens.

Maintenance Window Start (HH:MM) Window start in local server time (e.g. 03:00).

Maintenance Window End (HH:MM) Window end in local server time (e.g. 05:00).

Registration Security & ACL Alarm

Failed Registrations Threshold (per 15 min) Triggers a health check alarm when failed SIP authentication/registration attempts exceed this count in 15 minutes.

Live Diagnostics & Logs

Enables full PJSIP packet dumping to Asterisk logs. Automatically switches itself off after 10 minutes.

Rotates Asterisk messages and security log files without interrupting active calls.

Phone Settings

PBX Configuration Guide

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PBX Fine Tuning Guide

SIP Protocol Timers & Max-Forwards

Adaptive Jitter Buffer

Safe Night Maintenance Restart

Registration Security & ACL Alarm

Live Diagnostics & Logs

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