5 — PBX Features #

All PBX features write directly to FusionPBX / FreeSWITCH and take effect immediately — no restart required. Each feature requires the corresponding permission to be enabled on your account.

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Extensions #

Permission required: Extensions

Extensions are SIP endpoints (users/phones) registered in FreeSWITCH. Every device, ring group, queue agent, and voicemail box is associated with an extension.

List #

Go to PBX → Extensions.

The list shows extension number, display name, and tenant. Use the search bar to filter.

Add / Edit Extension #

Click Add Extension or the edit icon.

Field	Required	Description
Extension	✅	The SIP extension number (e.g. 1001)
Password	✅	SIP registration password
Display Name		Name shown on caller ID
Voicemail Password		PIN for voicemail access (default = extension number)
Enabled		Whether the extension is active

After saving, provision a SIP phone or softphone with:

• SIP Server: your ICTPBX server IP/hostname
• Username: the extension number
• Password: the SIP password set above

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Devices #

Permission required: Devices

Devices are physical or software SIP phones registered against extensions.

List #

Go to PBX → Devices.

Add / Edit Device #

Click Add Device or the edit icon.

Field	Required	Description
Device MAC	✅	MAC address of the physical phone (used for auto-provisioning)
Username	✅	SIP username (usually the extension number)
Template		Auto-provisioning template (e.g. Yealink, Polycom)
Enabled		Active/inactive

Lines tab: Assign the device to one or more extensions by selecting them in the Lines section.

Provisioning a Phone Step by Step #

This walkthrough registers a desk phone against an extension and gives it a config it pulls on reboot. The example uses a Grandstream GXP1628 on extension 101, but the flow is the same for any supported vendor.

Step 1 – Add the device. Open Devices from the left menu and click Add Device. Fill in the Device Information card:

Field	What to enter
MAC Address (required)	The phone’s MAC, for example c0:74:ad:e8:43:d7. Any format works; colons and dashes are stripped automatically. Must be 12 hex characters.
Label	A friendly name, for example “Randy’s Desk Phone”.
Vendor	Pick from the dropdown, for example Grandstream.
Model	Pick the model, for example GXP1628. This auto-fills the Template.
Template	Auto-filled from the model. Leave it as-is unless you know it differs.
Device Profile	Optional. Leave “No Profile” unless you have set one up.
Enabled	Keep Active checked.

The remaining fields (Firmware, Serial, Location, Username/Password, Description) are optional.

Tenant binding: the device ties to a tenant through the extension you assign in Step 2, because that extension belongs to a tenant domain. So pick an extension that belongs to the right customer.

Step 2 – Assign a SIP line. Stay on the same screen and scroll down to the SIP Lines card:

1. Click + Add Line. A row appears with Server set to demo.ictpbx.com and transport set to UDP. Those defaults are correct, so leave them.
2. Open the Extension dropdown and pick the extension, for example 101 – Randy. Selecting it auto-fills the User ID, Auth ID, and Display Name from that extension, including the SIP password. You never type credentials by hand.
3. Leave Server as demo.ictpbx.com and Transport as UDP.
4. The SIP port defaults to 5080 behind the scenes, the port FreeSWITCH listens on, so there is no field to set.

Step 3 – Save. Click Save. The form then shows a Provisioning URL, for example https://demo.ictpbx.com/provision/c074ade843d7.

Step 4 – On the phone (customer side). Set the Config Server Path to demo.ictpbx.com/provision over HTTPS, then choose Save and Reboot. The phone pulls its config and registers, exactly as extension 101 just did.

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Ring Groups #

Permission required: Ring Groups

Ring groups ring multiple extensions simultaneously or sequentially when a number is dialled.

List #

Go to PBX → Ring Groups.

Add / Edit Ring Group #

Field	Required	Description
Name	✅	Descriptive name
Extension	✅	The number callers dial to reach this group
Strategy	✅	simultaneous — all ring at once; sequence — one at a time
Timeout	✅	Seconds to ring before going to timeout destination
Timeout Destination		Where to send the call if unanswered (voicemail, IVR, etc.)
Members	✅	Extensions to include; each has its own ring timeout

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Call Queues #

Permission required: Call Queues

Call queues place callers in a waiting line, distributing them to available agents.

List #

Go to PBX → Call Queues.

Add / Edit Call Queue #

Field	Required	Description
Name	✅	Queue name
Extension	✅	Number callers dial to enter the queue
Strategy	✅	ring-all, longest-idle-agent, round-robin, top-down, agent-with-fewest-calls, random
Max Wait Time		Seconds before caller is sent to timeout destination
Max No-Answer		Times an agent can miss before being removed from the queue
Agents	✅	Extensions serving this queue; each can have a tier level and tier position
MOH Sound		Music on hold played while waiting
Timeout Destination		Where to route calls that exceed max wait time

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IVR Menus #

Permission required: IVR Menus

IVR (Interactive Voice Response) menus present callers with a recorded greeting and keypad options that route to other destinations.

List #

Go to PBX → IVR Menus.

Add / Edit IVR Menu #

Field	Required	Description
Name	✅	Menu name
Greeting	✅	Audio file played when callers enter
Invalid Sound		Audio played on wrong key press
Exit Sound		Audio played on exit
Timeout	✅	Seconds to wait for input before re-playing greeting
Exit Action		Destination if max retries exceeded
Options	✅	Key → Destination mappings (e.g. 1 → Sales ring group, 2 → Support queue, 0 → Operator extension)

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Voicemail #

Permission required: Voicemail

Voicemail boxes store recorded messages left by callers. Each box is linked to an extension.

List #

Go to PBX → Voicemail.

Add / Edit Voicemail Box #

Field	Required	Description
Extension	✅	The extension this mailbox belongs to
Password	✅	PIN to retrieve messages
Email		Address to receive message notifications (with attachment)
Greeting		Custom greeting audio file
Enabled		Active/inactive

Callers reach voicemail when their call is not answered and the extension forwards to voicemail (configured in the extension’s no-answer destination in FusionPBX).

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Conferences #

Permission required: Conferences

Conference rooms allow multiple callers to join a shared audio bridge by dialling an extension.

List #

Go to PBX → Conferences.

Add / Edit Conference #

Field	Required	Description
Name	✅	Room name
Extension	✅	Number callers dial to join
PIN		Optional entry PIN
Admin PIN		Moderator PIN (moderators can control the conference)
Max Members		Maximum simultaneous participants
Enabled		Active/inactive

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Music on Hold #

Permission required: Music on Hold

Music on Hold (MOH) categories hold audio files played to callers placed on hold or waiting in a queue.

List #

Go to PBX → Music on Hold.

Add / Edit MOH Category #

Field	Required	Description
Name	✅	Category name (referenced by queues and ring groups)
Rate		Audio sample rate (8000 Hz default)
Shuffle		Play files in random order
Files		Upload .wav or .mp3 audio files

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Follow Me #

Permission required: Follow Me

Follow Me forwards calls from an extension to one or more destinations in sequence or simultaneously — useful for mobile workers.

List #

Go to PBX → Follow Me.

Add / Edit Follow Me #

Field	Required	Description
Extension	✅	The source extension to forward from
Destinations	✅	Phone numbers or extensions to forward to; each has a timeout and delay
Strategy		simultaneous or sequence
Prompt		Whether to ask the receiving party to accept the call

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Call Flows #

Permission required: Call Flows

Call Flows are toggleable routing rules — a single extension can be switched between two destinations (e.g. normal hours → queue; after hours → voicemail). Useful for on/off-duty toggles.

List #

Go to PBX → Call Flows.

Add / Edit Call Flow #

Field	Required	Description
Name	✅	Descriptive name
Extension	✅	The number that triggers this flow
Status		Active (routes to Active Destination) or Inactive (routes to Inactive Destination)
Active Destination	✅	Where to route when the flow is active
Inactive Destination	✅	Where to route when the flow is inactive

Toggle the flow status between Active/Inactive to redirect calls on the fly.

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Time Conditions #

Permission required: Time Conditions

Time Conditions automatically route calls based on day/time, enabling business-hours routing without manual intervention.

List #

Go to PBX → Time Conditions.

Add / Edit Time Condition #

Field	Required	Description
Name	✅	Descriptive name
Extension	✅	The number this condition is applied to
Match Destination	✅	Where to route when within the time window
No-Match Destination	✅	Where to route outside the time window
Time Rules	✅	One or more time windows: days of week, start time, end time

Multiple time rules are OR-combined — the condition matches if any rule matches the current time.

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Call Block #

Permission required: Call Block

Call Block allows you to reject calls from specific numbers or number patterns.

List #

Go to PBX → Call Block.

Add / Edit Block Rule #

Field	Required	Description
Number	✅	The number to block (exact match or pattern)
Action	✅	Hangup, Busy, or Play message
Description		Note about why this number is blocked
Enabled		Active/inactive

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Inbound Routes #

Permission required: Inbound Routes

Inbound Routes map inbound DID numbers to internal destinations (extensions, ring groups, IVR menus, etc.). This is the first routing decision made when a call arrives from the carrier.

List #

Go to PBX → Inbound Routes.

Add / Edit Inbound Route #

Field	Required	Description
DID Number	✅	The inbound phone number from your carrier
Caller ID Name		Optional display name
Destination	✅	Where to route this number (extension, IVR, ring group, etc.)
Enabled		Active/inactive

After saving, FreeSWITCH is automatically updated to route calls to that DID to the selected destination.

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Gateways #

Gateways (SIP Trunks) connect ICTPBX to your telecom carrier for outbound and inbound calls.

List #

Go to PBX → Gateways.

Add / Edit Gateway #

Field	Required	Description
Name	✅	Unique gateway name (used in FreeSWITCH config)
Username	✅	SIP username provided by your carrier
Password	✅	SIP password
Proxy / Realm	✅	Carrier SIP server address
From Domain		SIP domain in From header
Register		Whether to register with the carrier (true for most hosted trunks)
Fax Support		Enables T.38 fax codec negotiation on this trunk
Enabled		Active/inactive

After saving, ICTPBX:

1. Creates/updates the gateway record in FusionPBX
2. Writes the SIP profile XML to disk (/etc/freeswitch/sip_profiles/external/)
3. Reloads the FreeSWITCH external profile — the gateway goes REGED within seconds