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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.

FreePBX Trunks

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

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>

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

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"

FreePBX chan_pjsip configure pjsip advanced settings Expiration

FreePBX chan_pjsip configure pjsip advanced settings Force rport

Codecs​

Configure the "Codecs" settings in the following order:

  • G722
  • alaw
  • ulaw

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".

FreePBX Asterisks SIP settings

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

FreePBX/Asterisk activate keep alive

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

FreePBX save changes

Further Information​

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

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