The Conferences module in FreePBX creates audio meeting rooms where multiple callers can join the same bridge. Modern FreePBX uses Asterisk ConfBridge under the hood (stable, flexible, and production-ready).

In real deployments, conferences are used for:

• Internal team meetings (daily standups, supervisor coaching)
• Live customer escalation (agent + supervisor + customer)
• Training rooms (listen-only participants)
• AI-to-human handoff (MYLINEHUB VoiceBridge can transfer/bridge a call into a conference room)

Important concept: A conference is a destination. You can route calls to it from an IVR, inbound route, ring group, queue, time condition, or custom dialplan.

Prerequisites Before Creating a Conference

• Extensions are working and can call each other with two-way audio
• DTMF works reliably (IVR digit presses are detected correctly)
• Firewall/NAT/RTP are stable (one-way audio issues are solved first)

Conferences amplify RTP/codec problems. If your basic extension-to-extension call is not stable, fix networking first before tuning conference options.

Where to Find Conferences in FreePBX

Applications → Conferences

Screenshot 1 — Conference Edit Screen (ConfBridge Options)

At the top you may see a banner like “Used as Destination by …”. That means this conference room is referenced elsewhere (IVR/route/queue). If you delete or renumber it, those call flows can break—so treat conference numbers like production destinations.

Core Fields Explained (As Shown in the Screen)

Conference Number

• The internal dialable number for this room (example: 1234).
• Must be unique and not conflict with extensions or feature codes.
• In call flows: IVR / inbound route can send calls to this number as a destination.

Conference Name

• Human-friendly label (example: Main Conference).
• Used for admin clarity and some UI displays; callers still dial the conference number.

User PIN and Admin PIN

• User PIN: participants enter this to join as a normal user.
• Admin PIN: moderators/leaders enter this to join with elevated permissions.

Production best practice: Always set a strong Admin PIN. If you want open internal joining, you can leave User PIN empty for internal-only access, but do not do this for internet-exposed systems.

Language

• Controls system prompts language for conference messages (join/leave, menu prompts, etc.).
• Use Inherit unless you have multi-language prompt packs and want per-room behavior.

Join Message

• Audio played when someone joins (often set to None for low-noise call centers).
• For meetings, you may enable prompts so users know they successfully joined.

Leader / Moderator Behavior

Leader Wait

• If enabled, participants will wait until a leader/moderator joins (usually via Admin PIN).
• Useful for meetings where the host should arrive first.

Leader Leave

• Controls what happens when the leader leaves.
• Common policy: end the conference when leader leaves (prevents “unmoderated room”).

Call center tip: For escalation bridges, you usually do not want the whole room to drop if one supervisor disconnects. Decide this based on your real workflow.

Talker Options, Quiet Mode, and Announcements

Talker Optimization

• Optimizes audio processing when many participants are present.
• Generally safe to enable in larger rooms; keep defaults unless you have a reason to change.

Talker Detection

• Detects who is speaking (used for talker announcements or certain conference features).
• Can add processing overhead; in noisy environments you may keep it OFF unless needed.

Quiet Mode

• Reduces spoken system announcements to keep the room quiet.
• Good for call-center style bridges or when join/leave noise is distracting.

User Count

• If enabled, the system can announce how many users are in the conference.
• Useful for meetings; often disabled for customer-facing scenarios.

User Join/Leave

• Controls join/leave notifications (beeps/announcements depending on system settings).
• Enable for meeting awareness; disable for minimal noise in production call flows.

Music on Hold and Menu Options

Music on Hold

• If enabled, callers may hear music in waiting conditions (commonly used with leader-wait).
• If leader-wait is enabled, MOH makes the waiting experience better.

Music on Hold Class

• Selects which MOH playlist/class to use (often inherit is fine).
• In production you can use a custom MOH class for branded music or compliance announcements.

Allow Menu

• If enabled, participants can use ConfBridge/FreePBX in-conference menu features (DTMF controls).
• Example use-cases: mute/unmute self, hear participant count, admin controls.

Security note: If you enable menus, ensure roles/PINs are set correctly so normal users do not get admin controls.

Recording and Participant Controls

Record Conference

• If enabled, the conference audio can be recorded.
• Great for QA, training review, compliance, and AI analytics—only if your legal/privacy policy allows it.

Maximum Participants

• Limits how many users can join (in the screen, 0 typically means “no limit”).
• Set a realistic limit to avoid CPU/network overload on small servers.

Mute on Join

• If enabled, participants join muted by default.
• Useful for training rooms, webinars, or large internal meetings.

Member Timeout

• Maximum time a participant can stay connected before timeout (seconds).
• Use carefully—too small can drop legitimate long meetings; too large is usually fine.

Production Recommendations (Simple Defaults)

• Set Admin PIN strong; set User PIN based on whether this is internal-only or customer-facing.
• Quiet Mode: enable for call-center bridges; disable for meetings if you want announcements.
• User Join/Leave: enable for meetings; disable for customer flows.
• Leader Wait: enable for meetings; optional for escalation bridges.
• Record Conference: enable only if policy allows and storage is planned.
• Max Participants: set a safe cap for your server size.

Troubleshooting (Most Common Issues)

No audio / one-way audio

• RTP ports blocked in firewall
• NAT/external IP settings incorrect
• Remote endpoints behind strict NAT without keepalive

PIN not detected

• DTMF mode mismatch (RFC4733/RFC2833 vs inband)
• Codec/transcoding issues affecting DTMF

Participants drop randomly

• UDP timeouts on router/firewall
• SIP session timers / NAT rebinding
• Network instability

If you need deeper debugging, use Asterisk CLI:

asterisk -rvvv
pjsip set logger on
rtp set debug on

MYLINEHUB Usage Note (Practical)

A conference room can be used as a clean escalation bridge:

• AI/IVR answers the call
• VoiceBridge decides escalation needed
• Call is routed/transferred into a conference number
• Supervisor/agent joins with Admin PIN (or a dedicated internal rule)

This approach is simple, stable, and avoids complex multi-bridge edge cases when you just need “everyone in one room”.

Next Steps

1. FreePBX Extensions (PJSIP)
2. FreePBX IVR
3. FreePBX Queues
4. FreePBX Firewall