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FreePBX

This guide explains how to set up a sipcall trunk or a sipcall phone number with FreePBX.

note

This guide refers to FreePBX version 15 and above.

Configuration​

Add Trunk​

Navigate to the "Connectivity" menu and select "Trunks" from the dropdown menu.

Image: FreePBX Trunks

Click "Add Trunk" and select "Add SIP (chan_pjsip) Trunk".

Image: FreePBX chan_pjsip add trunk

Set a "Trunk Name" and the "Outbound CallerID" to your trunk's main number as shown in login.sipcall.ch

  • Trunk Name: <freely chosen name for your trunk>
  • Outbound CallerID: <userID>

Image: FreePBX chan_pjsip configure general tab

Registration​

note

sipcall offers various SIP servers. More information can be found here.

Configure the "General" settings under the "pjsip Settings" tab as shown below:

  • Username: <userID>
  • Auth username: <userID>
  • Password: <password>
  • SIP Server: <SIP server>
  • SIP Server Port: 5060
  • Context: from-pstn-toheader

Image: FreePBX chan_pjsip configure pjsip settings

Advanced Settings​

Configure the "Advanced" settings as shown below:

  • DTMF Mode: RFC 4733
  • Expiration: 60 (for 60 seconds)
  • Force rport: select "No"

Image: FreePBX chan_pjsip configure pjsip advanced settings Expiration

Image: FreePBX chan_pjsip configure pjsip advanced settings Force rport

Codecs​

Configure the "Codecs" settings in the following order:

  • G722
  • alaw
  • ulaw

Image: FreePBX chan_pjsip codecs

RTP Keep Alive​

To prevent audio stream problems, please enable RTP Keep Alive. This option can be found in the "Settings" tab under "Asterisk SIP Settings".

Image: FreePBX Asterisks SIP settings

Change the "RTP Keep Alive" setting from "0" to "1".

Image: FreePBX/Asterisk activate keep alive

To save the settings, click "Submit" and then "Apply Config" in the top left corner.

Image: FreePBX save changes

Further Information​

Tips and tricks for FreePBX can be found in the FreePBX Tips and Tricks section.

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