You can integrate your corporate SIP PBX into TrueConf Server via the built-in multiple gateway. This guide will show you how to connect TrueConf Server with Asterisk, a PBX with FreePBX 12 GUI.
Table of Contents
Step 1: Create a SIP Trunk on the Asterisk Side
- In the PBX control panel, go to Connectivity → Trunks.
- Choose Add a SIP (chan_sip) trunk in the opened window:
- On the Add a trunk page:
- Enter the trunk name and outgoing CallerID.
- In Options for PEER, enter:
- CallerID (see Item A).
- Trunk password.
- IP address of your TrueConf Server instance.
- To save changes in your PBX settings, click Apply Changes.
Step 2: Set up TrueConf Server
- Go to the TrueConf Server control panel.
- In the Gateways→ SIP section, click Add Configuration.
- Choose manual configuration of SIP connection and click Create.
- To add a PBX connection rule fill in the form by entering:
- Asterisk PBX address (e.g. asterisk.company.ru) and its port in the Host and Port fields.
- CallerID and SIP trunk password (see Step 1) in the Login and Password fields.
In addition, check the Default VoIP Server box.
Step 3: Call a PBX from TrueConf Client Application
By now, setup is complete, and you can call PBX subscribers using the dialer in our application or by finding a subscriber’s number in the address book. Use the call string with #sip: and #tel: prefixes to identify PBX subscribers when administering the server (creating aliases, requesting API, compiling the address book, etc.).
PBX subscribers can call back, if the server connection rule was added to PBX settings.
