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Paging Groups & Intercom Module Documentation

15 min readUpdated: Sep 26, 2026
View as Markdown
  1. Module Overview (Technical)
  2. Module Overview (Commercial/Business)
  3. Module Overview (End User/Administrator)
  4. Configuration Fields Reference
  5. Call Flow / Logic Explanation
  6. Common Scenarios & Examples
  7. Model Context Protocol (MCP) AI Integration
  8. Limitations & Important Notes
  9. Troubleshooting Tips
  10. Glossary

Paging Groups provide one-way or two-way audio broadcasting to multiple endpoints simultaneously. This includes:

  • Paging: One-way announcements to groups of phones
  • Intercom: Two-way communication (duplex mode)
  • Multicast: Network-level audio broadcasting to compatible devices
Mode Description Use Case
Default Standard SIP paging to selected extensions Office announcements
Duplex Two-way intercom communication Warehouse communication
Multicast RTP multicast to IP speakers Factory floor PA
Hybrid Both SIP extensions and multicast Mixed environments
┌─────────────────────────────────────────────────────────────────┐
│ Paging System Architecture │
├─────────────────────────────────────────────────────────────────┤
│ │
│ User dials: *80 (Paging Group Code) │
│ │ │
│ ▼ │
│ ┌──────────────────────────────────────────────────────────┐ │
│ │ Paging Group Lookup │ │
│ │ SELECT * FROM paging_groups WHERE code = '*80' │ │
│ │ → Returns mode, members, settings │ │
│ └──────────────────────────────────────────────────────────┘ │
│ │ │
│ ▼ │
│ ┌──────────────────────────────────────────────────────────┐ │
│ │ Based on Mode │ │
│ │ │ │
│ │ DEFAULT ────────────────────────────────────────────── │ │
│ │ │ │ │
│ │ ├─ Create conference room │ │
│ │ ├─ Set auto-answer headers │ │
│ │ ├─ Call each member extension │ │
│ │ └─ Bridge caller to conference │ │
│ │ │ │
│ │ DUPLEX (Intercom) ──────────────────────────────────── │ │
│ │ │ │ │
│ │ ├─ Same as default │ │
│ │ └─ Enable two-way audio for all participants │ │
│ │ │ │
│ │ MULTICAST ──────────────────────────────────────────── │ │
│ │ │ │ │
│ │ ├─ Answer caller │ │
│ │ └─ Stream audio to multicast address (239.x.x.x) │ │
│ │ │ │
│ │ HYBRID ─────────────────────────────────────────────── │ │
│ │ │ │ │
│ │ ├─ Call SIP extensions (like default) │ │
│ │ └─ Simultaneously stream to multicast │ │
│ │ │ │
│ └──────────────────────────────────────────────────────────┘ │
│ │
└─────────────────────────────────────────────────────────────────┘

Paging Groups enable instant mass communication:

External Systems Built-In Paging
Separate PA system Integrated with PBX
Additional hardware Uses existing phones
Manual intercom setup Configured via UI
Limited scalability Unlimited groups
  1. Office Announcements

    • “John Smith, please come to reception”
    • Emergency notifications
    • Company-wide messages
  2. Warehouse/Manufacturing

    • Foreman announcements
    • Safety alerts
    • Shift change notifications
  3. Intercom Communication

    • Reception to back office
    • Security to all stations
    • Medical triage coordination
  4. School/Campus

    • Classroom announcements
    • Emergency lockdown
    • Bell system integration
Feature Benefit
Auto-Answer Phones answer automatically—no user action needed
Mute Participants One-way paging (listeners can’t interrupt)
PIN Protection Restrict who can initiate pages
Recording Record all announcements for compliance
Multicast Broadcast to IP speakers without SIP overhead
Skip Busy Don’t interrupt ongoing calls

3. Module Overview (End User/Administrator)

Section titled “3. Module Overview (End User/Administrator)”
  • Create paging groups with assigned extension codes
  • Add/remove member extensions
  • Configure one-way (paging) or two-way (intercom) mode
  • Set up multicast for IP speakers
  • Protect pages with PIN codes
  • Enable recording of announcements
  1. Navigate to PBX → Applications → Paging & Intercom in the main sidebar.
  2. The list view displays all configured paging groups, detailing their dial Code, Name, Members count, Mode/Group Strategy, and active Status (Enabled/Disabled).
  3. Click the + Add button in the upper toolbar to configure a new paging group.
  4. Click any row or the edit icon to configure group extensions, broadcast strategies, auto-answer headers, or timeouts.

Paging & Intercom List View

┌─────────────────────────────────────────────────────────────────┐
│ Creating a Paging Group │
├─────────────────────────────────────────────────────────────────┤
│ │
│ Step 1: Basic Settings │
│ ├─ Code: 7001 (or *80) │
│ ├─ Group Name: "Warehouse Overhead Paging" │
│ ├─ Strategy: Default (SIP auto-answer) │
│ └─ Timeout: 15 seconds │
│ │
│ Step 2: Add Members │
│ ├─ Search: "Warehouse" or "Floor" │
│ ├─ Select extensions from the searchable list │
│ └─ Counter shows "X of Y selected" │
│ │
│ Step 3: Caller ID Display │
│ ├─ Name Template: Paging ${extension_name} │
│ └─ Number Template: ${extension_number} │
│ │
│ Step 4: Enable & Save │
│ └─ Users dial 7001 → Member phones beep and auto-answer │
│ │
└─────────────────────────────────────────────────────────────────┘

[!TIP] Auto-Answer Required: For paging to work, recipient IP phones must support and permit auto-answer SIP headers (Alert-Info: info=alert-autoanswer or Call-Info: answer-after=0).

[!TIP] Skip Busy: When configuring large teams, skip busy extensions so active customer calls are not interrupted by overhead broadcasts.

[!CAUTION] Multicast Network Config: Multicast paging requires proper LAN switch configuration (IGMP snooping and multicast routing) if speakers reside on different VLANs.


Paging & Intercom Configuration Form

Field Technical Description User-Friendly Tooltip Example Required Notes
Code * Dialable extension or feature code for this paging group Digits dialed by authorized users to initiate an announcement broadcast 7001, *80 Yes Must be unique within the domain dialplan registry.
Group Name * Human-readable identifier for the paging group Clear label describing the department or area targeted Warehouse Overhead Paging, Executive Intercom Yes Displayed in directories and audit logs.
Group Strategy (Mode) Broadcast methodology and audio transmission mode Operational audio mode (Default [One-way SIP], Duplex [Two-way intercom], or Multicast [RTP multicast]) Default Yes FreeSWITCH conference-driven auto-answer.
Multicast Address IP multicast address and UDP port for streaming audio directly to network hardware speakers Multicast address and port (active when Mode is Multicast) 239.255.10.10:5000 Conditional Required only when Mode is set to Multicast.
Timeout (seconds) * Maximum duration before automatically terminating page Time limit in seconds for the broadcast session (5 - 120s) 15, 30 Yes FreeSWITCH leg_timeout. Protects against un-hung channels.
Caller ID Name Display name presented to called endpoints Caller ID name presented on receiving phone screens Overhead Page No Overrides originator display name.
Caller ID Number Display number presented to called endpoints Caller ID number presented on receiving phone screens 7001 No Overrides originator extension number.
Enabled Master activation toggle Instantly enable or disable this paging group Yes / No (toggle) Yes Default: Yes.
Members (Extensions) Multi-select list of extensions included in this broadcast group Select which phone extensions will auto-answer when this group is called Multi-selection list Yes Recipients receive native sip_auto_answer=true.
Field Technical Description User-Friendly Tooltip Default Example Notes
Audio Codec RTP audio codec used for the broadcast session Preferred audio codec for the stream PCMU PCMU, PCMA, G722, OPUS Multicast speakers must support the selected codec.
Speaker Volume Audio output volume adjustment applied to broadcast legs Speaker volume adjustment (-5 to +5) 5 0 (normal), 5 (loud) Sets FreeSWITCH speak_volume leg channel variable.
Pre-Page Announcement Audio recording played to caller/listeners before bridging Audio prompt played prior to the live broadcast None beep.wav, chime.wav Uploaded via Sound Prompts / Voice Recordings.
Record Session Records the complete paging session to storage Enable or disable automatic recording of announcements Off Toggle (On/Off) Saves audio via FreeSWITCH record_session=true.
Skip Busy Extensions Prevents interrupting active calls on recipient endpoints Do not ring or auto-answer extensions currently on an active call Off Toggle (On/Off) FreeSWITCH skip_busy=true originate channel variable.

┌─────────────────────────────────────────────────────────────────┐
│ Standard Paging Flow │
├─────────────────────────────────────────────────────────────────┤
│ │
│ 1. User dials *80 (paging group code) │
│ │ │
│ ▼ │
│ 2. Check PIN requirement (if configured) │
│ ├─ PIN required → Prompt for PIN │
│ │ ├─ Correct → Continue │
│ │ └─ Wrong → "Invalid PIN" + hangup │
│ └─ No PIN → Continue │
│ │ │
│ ▼ │
│ 3. Play pre-announcement (if configured) │
│ │ │
│ ▼ │
│ 4. Create dynamic conference for this page │
│ │ │
│ ▼ │
│ 5. For each member extension: │
│ ├─ Check if busy (if skip_busy enabled) │
│ ├─ Set auto-answer SIP headers │
│ └─ Bridge to conference (muted if mute_participants) │
│ │ │
│ ▼ │
│ 6. Wait for specified timeout for endpoints to answer │
│ │ │
│ ▼ │
│ 7. Caller speaks → Audio broadcast to all answered phones │
│ │ │
│ ▼ │
│ 8. Caller hangs up → Conference ends, all legs disconnect │
│ │
└─────────────────────────────────────────────────────────────────┘
┌─────────────────────────────────────────────────────────────────┐
│ Multicast Paging Flow │
├─────────────────────────────────────────────────────────────────┤
│ │
│ 1. User dials *85 (multicast paging group) │
│ │ │
│ ▼ │
│ 2. Answer caller │
│ │ │
│ ▼ │
│ 3. Start RTP stream to multicast address: │
│ Address: 239.255.10.10:5000 │
│ Codec: PCMU (G.711µ) │
│ │ │
│ ▼ │
│ 4. IP speakers (subscribed to 239.255.10.10) receive audio │
│ │ │
│ ▼ │
│ 5. Caller hangs up → Stream stops │
│ │
└─────────────────────────────────────────────────────────────────┘

Setup:

Setting Value
Code *80
Name All Office
Mode Default
Members 1001-1050 (all office extensions)
Auto Answer ✓
Mute Participants ✓
Skip Busy ✓

Result: Dial *80, all available phones auto-answer (muted), announcement plays.

Setup:

Setting Value
Code *70
Name Warehouse Intercom
Mode Duplex
Members 2001, 2002, 2003 (warehouse phones)
Intercom Mode ✓
Mute Participants ✗ (allow responses)

Result: Dial *70, all warehouse phones connect for two-way conversation.

Setup:

Setting Value
Code *85
Name Factory PA
Mode Multicast
Multicast Address 239.255.10.10
Port 5000
Codec PCMU

Result: Dial *85, audio streams to all IP speakers configured for that multicast group.

Scenario 4: Protected Emergency Announcement

Section titled “Scenario 4: Protected Emergency Announcement”

Setup:

Setting Value
Code *911
Name Emergency Broadcast
Mode Hybrid
PIN Code 9911
Members ALL extensions
Multicast (also to PA speakers)
Recording ✓

Result: Only authorized users with PIN can trigger emergency broadcast to all phones and speakers. All broadcasts recorded.


7. Model Context Protocol (MCP) AI Integration

Section titled “7. Model Context Protocol (MCP) AI Integration”

Paging and Intercom groups are exposed via the Model Context Protocol (MCP) for programmatic administration, membership audits, and direct voice broadcast origination by authorized AI operators.

Tool Name Operation Description Access Level
list_paging_groups Query List all paging and intercom groups, audio modes, member counts, and multicast addresses. Read-Only
get_paging_group_status Query Retrieve full group details including individual member extensions and audio parameters. Read-Only
create_paging_group Provisioning Create a new paging or two-way intercom group with auto-answer headers and custom timeouts. Admin / Superadmin
update_paging_group Management Add/remove member extensions, change broadcast audio mode, or adjust timeout limits. Admin / Superadmin
delete_paging_group Deprovisioning Remove a paging group from the PBX. Superadmin
page_broadcast Live Telephony Trigger a real-time live page broadcast originating from a specific extension to a paging group code. Operator / Admin

Provisions an auto-answer broadcast group within the domain.

{
"name": "create_paging_group",
"description": "Create a new Paging / Intercom group to broadcast voice messages or execute two-way intercom to multiple extensions with auto-answer.",
"inputSchema": {
"type": "object",
"properties": {
"code": { "type": "string", "description": "Dial code for the paging group (e.g. '*84', '800', '99')." },
"name": { "type": "string", "description": "Name of the paging group (e.g. 'Warehouse Speakers')." },
"members": {
"type": "array",
"items": { "type": "string" },
"description": "Array of member extension numbers (e.g. ['2001', '2002', '2003'])."
},
"mode": {
"type": "string",
"enum": ["default", "intercom", "multicast", "duplex"],
"description": "Paging mode: 'default' (one-way), 'intercom' (two-way), 'multicast' (RTP stream), 'duplex' (conference)."
},
"timeout": { "type": "number", "description": "Maximum broadcast duration in seconds (default: 30)." },
"autoAnswer": { "type": "boolean", "description": "Send SIP auto-answer headers (Call-Info / Alert-Info)." }
},
"required": ["code", "name", "members"]
}
}
  1. Domain Number Collision Prevention: The service enforces validateNumberUniqueness(code, domain.id, 'paging_groups') against public.dialplan_registry. Paging codes cannot collide with Extensions, Conferences, Parking Lots, Speed Dials, or Feature Codes in the same domain.
  2. Auto Dialplan Synchronization: Changes immediately sync to public.dialplan_registry and reload Telephony Server XML configuration so endpoints can dial the paging code without server restart.

Example 1: Creating a Warehouse Intercom Group

Section titled “Example 1: Creating a Warehouse Intercom Group”

Admin Prompt:
*“Create a two-way intercom group with code 85 named ‘Warehouse Floor’ for extensions 3010, 3011, and 3012 with a 60-second timeout.”

AI Tool Execution:

{
"tool": "create_paging_group",
"arguments": {
"code": "*85",
"name": "Warehouse Floor",
"members": ["3010", "3011", "3012"],
"mode": "intercom",
"timeout": 60,
"autoAnswer": true
}
}

MCP Response:

{
"success": true,
"data": {
"code": "*85",
"name": "Warehouse Floor",
"mode": "intercom",
"memberCount": 3,
"message": "Paging group *85 created successfully."
}
}

[!WARNING] Auto-Answer Support: Not all SIP phones support auto-answer. Older or basic phones may not activate.

[!WARNING] Multicast Network Requirements: Multicast requires IGMP-enabled switches and proper network configuration.

[!IMPORTANT] Codec Compatibility: Multicast endpoints must support the configured codec. PCMU (G.711µ) has widest support.

  1. Test Auto-Answer: Verify phones actually auto-answer before relying on paging
  2. Network Planning for Multicast: Work with network team to enable IGMP snooping
  3. PIN Protection: Use PINs for high-impact pages (emergency, company-wide)
  4. Skip Busy: Always enable to avoid interrupting important calls
  5. Keep Groups Focused: Create smaller, targeted groups rather than one huge group

For auto-answer to work, phones typically need:

  • Auto-answer header support enabled
  • Intercom mode enabled in phone settings
  • Correct SIP headers: Alert-Info: info=auto-answer or similar

Symptom Possible Cause Solution
Phone doesn’t answer Auto-answer not supported Check phone model/settings
No audio on page Codec mismatch Verify codec compatibility
Multicast not working Network issue Check IGMP, switch config
Only some phones answer Skip busy + busy phones Expected behavior
“Invalid PIN” Wrong PIN entered Verify correct PIN
Recording not created Path permissions Check directory permissions

List paging groups:

SELECT pg.code, pg.name, pg.mode, pg.enabled,
COUNT(m.id) as member_count
FROM public.paging_groups pg
LEFT JOIN public.paging_group_members m ON pg.id = m.paging_group_id
WHERE pg.domain_id = [domain_id]
GROUP BY pg.id
ORDER BY pg.code;

Check members of a group:

SELECT pgm.extension, e.description
FROM public.paging_group_members pgm
JOIN public.sip_extensions e ON pgm.extension = e.extension
WHERE pgm.paging_group_id = [group_id];

Check active conferences (paging uses conferences):

Terminal window
fs_cli -x "conference xml_list"

Debug multicast:

Terminal window
# Check if multicast traffic is being sent
tcpdump -i eth0 host 239.255.10.10 and udp port 5000

Term Definition
Paging One-way audio broadcast to multiple phones
Intercom Two-way audio communication between phones
Multicast Network-level broadcast using IP multicast (239.x.x.x addresses)
Auto-Answer Phone feature that answers incoming calls automatically
Duplex Two-way audio communication mode
IGMP Internet Group Management Protocol—enables multicast on switches
TTL Time To Live—how many network hops multicast packets can travel
PCMU G.711µ audio codec—widely supported, good quality
RTP Real-time Transport Protocol—carries audio over IP networks
PA System Public Address system—speakers for announcements
Barge-In Ability to interrupt an ongoing announcement
Skip Busy Don’t page extensions that are on active calls

Documentation last updated: January 2026