Note: IP PBX replaces the Linked Office feature.
The IP PBX Connector enables integration between a third-party IP PBX system (e.g. 3CX, Asterisk) and uPBX. This is useful in situations such as:
- Acquiring a new business and temporarily integrating phone systems
- Migrating from an existing IP PBX to uPBX
- Custom call routing or hybrid environments
Connector Configuration
On the left-hand side of the screen, you can configure the connector details:
| Field | Description |
|---|---|
| Connector Type | Currently only IP PBX is supported. |
| Name | A name to identify this connector (e.g. "Demo", "Sydney Office PBX"). |
| Country | Used for formatting numbers and dialling rules. |
| Timezone | Determines the local time context for logs and call flow operations. |
| Default Outbound Number | The number used for outbound calls when none is specified explicitly by the calling extension. |
| SIP Host | The SIP host address that your third-party PBX will connect to. |
| Authentication ID | A unique ID assigned by the system, used as a username to authenticate the trunk connection. |
| Password | The associated password for authentication. |
Note: The Authentication ID and Password are generated when a new connector is created. Use these credentials to configure a SIP trunk on your third-party PBX pointing to the SIP Host.
User Mapping
On the right-hand side of the screen, define the list of Users (Extensions) that will be accessible from your third-party PBX:
| Field | Description |
|---|---|
| Extension | Internal extension number (e.g. 880, 881). |
| Label / Name | An optional name/label for clarity (e.g. "Reception", "Support"). |
| Action | Allows you to delete an existing user from the list. |
Adding a User
To add a user:
- Enter an extension number in the Extension field (e.g.
891) - Click Add User
- Optionally, enter a label for the extension
- Click Save Users
These users can then be referenced in call flows, ring groups, and call queues within uPBX.
Example: 3CX Configuration
To connect 3CX to uPBX, configure a SIP Trunk.
Steps
- Log in to the 3CX Admin Console
- Go to SIP Trunks → Click Add SIP Trunk
- Choose:
- Country:
Generic - Provider:
Generic SIP Trunk - Main Trunk Number: your allocated number (e.g.
0285070701) - Click OK
Trunk Settings
| Setting | Value |
|---|---|
| Registrar/Server Hostname | demo.mycloudpbx.com.au |
| Number of SIP Channels | 10 (adjust as needed) |
| Authentication ID (SIP User ID) | 123456 (example) |
| Authentication Password | your_password_here |
| Outbound Proxy | (Leave blank unless advised) |
| DID/DDIs | Add your main number (e.g. 0285070701) |
- Configure Outbound Rules to use the new trunk.
- Map Inbound Rules to ring groups, auto attendants, or extensions as required.
Example: Asterisk Configuration
To connect Asterisk to uPBX, add a SIP trunk in pjsip.conf (or sip.conf if using chan_sip).
pjsip.conf
[mycloudpbx] type=endpoint transport=transport-udp context=from-external disallow=all allow=ulaw outbound_auth=mycloudpbx aors=mycloudpbx from_domain=demo.mycloudpbx.com.au [mycloudpbx] type=aor contact=sip:demo.mycloudpbx.com.au [mycloudpbx] type=auth auth_type=userpass username=123456 password=your_password_here
extensions.conf
[outbound-mycloudpbx]
exten => _0X.,1,Dial(PJSIP/${EXTEN}@mycloudpbx)
[from-external]
exten => 880,1,Dial(SIP/1001) ; Example route to local SIP user
Replace 785217 and your_password_here with the values provided in your connector settings.
Firewall and NAT Notes
*Ensure your firewall allows SIP (UDP 5060) and RTP (typically UDP ports 10000–20000).
*If your PBX is behind NAT, enable NAT traversal settings
*Always test both inbound and outbound calls for audio flow and call completion.