- 1 Requirements
- 2 General information
- 3 Test environment
- 4 Basic configuration
- 5 Trunk creation
- 6 Extensions
- 7 Inbound Routes
- 8 Outbound Routes
- 9 Terminal configuration
- peoplefone account (registration)
- FreePBX (Download)
- FreePBX Installation
- SIP Trunk and Clip Open (Formular) activation
- SIP Informations
- VoIP Client Software
In this article, we will explain how you can configure a trunk and an administration line to peoplefone on the FreePBX. For this you need access to the web interface of your FreePBX. With two phones (VoIP phones, hardware phones), you can test the configuration of your telephone system. To enable incoming calls to be tested, you need one or more target numbers, which you can either purchase from your customer account (buy national numbers) or request via the support.
This guide was created based on a FreePBX 32bit and 64bit Full installation version 10.13.66 with Asterisk 11. Since Asterisk 13 is still faulty (as of 09.12.2016), this should be used only for test purposes and not operationally.
We recommend using the FreePBX behind a firewall for security reasons. For this reason, we disabled the internal firewall of the FreePBX, switched off NAT and assigned the public IP address.
The FreePBX installation was done as follows:
- With a fixed internal IP address (IPv4)
- IPv6 has been deactivated
- Default Network (local / 24)
- Default Gateway (firewall)
- DNS (Google)
- Recommended firewall settings
Images STABLE FreePBX Linux 6.6 • Asterisk 11 or 13
10.13.66-64bit / Release Date: 2016
10.13.66-32bit / Release Date: 2016
General SIP Settings
FreePBX Webinterface → Settings → Asterisk SIP Settings → General SIP Settings
- Audio Codecs order alaw, ulaw, g722, g729
Chan SIP Settings
- Set NAT to no
- IP Configuration Public IP
FreePBX Webinterface → Connectivity → Trunks → Add Trunk
- Add a «SIP (chan_sip) trunk».
- Define your brand name
Trunk Dialed Number Manipulation Rules
- For match pattern, use «.»
Trunk SIP Settings Outgoing
- Define a trunk name and specify the PEER details
- Set the PEER details
Trunk SIP Settings Incoming
- In the USER context, enter the SIP username
FreePBX Webinterface → Applications → Extensions → Add Extension
- Add a new Chan_SIP extension
FreePBX Webinterface → Applications → Extensions → General
- Add a user extension
- Set the display name for this extension
- Set an outbound CID / phone number
- A secret / password is automatically generated for you, this is required during the terminal configuration (SNOM Web Interface) and entered in the Password field.
FreePBX Webinterface → Connectivity → Inbound Routes → General
- Add an inbound route
- Define a name for the inbound route
- Define a destination where the call is to be delivered.
Outbound Route Settings
- Define a name for the outbound route
Outbound Dial Patterns
- For match pattern, use «X.».
FreePBX Webinterface → Connectivity → Outbound Routes → Dial Patterns
- Define which trunk the connection should be made.
- As display name, you can enter something which is then displayed on the devices.
- For an account, you must enter the user extension.
- Use the automatically generated "Secret"
- At the Registrar you enter the domain or the IP address of your telephone system, the port must also be entered.
- As codec, please place the following in the same order: pcma, pcmu, g722, g729, telephone-event
- RTP Encryption is not supported and must be turned off.