FreePBX
This guide explains how to set up a sipcall trunk or a sipcall phone number with FreePBX.
This guide refers to FreePBX version 15 and above.
Configuration
Add Trunk
Navigate to the "Connectivity" menu and select "Trunks" from the dropdown menu.
Click "Add Trunk" and select "Add SIP (chan_pjsip) 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>
Registration
sipcall offers various SIP servers. More information can be found here.
Configure the "General" settings under the "pjsip Settings" tab as shown below:
- SIP Server
- SIP Server with DNS-SRV
- Username:
<userID> - Auth username:
<userID> - Password:
<password> - SIP Server:
<SIP server> - SIP Server Port:
5060 - Context:
from-pstn-toheader
- Username:
<userID> - Auth username:
<userID> - Password:
<password> - SIP Server:
<SIP server (DNS-SRV)> - SIP Server Port:
<leave empty> - Context:
from-pstn-toheader
Advanced Settings
Configure the "Advanced" settings as shown below:
- DTMF Mode:
RFC 4733 - Expiration:
60(for 60 seconds) - Force rport: select "No"
Codecs
Configure the "Codecs" settings in the following order:
- G722
- alaw
- ulaw
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".
Change the "RTP Keep Alive" setting from "0" to "1".
To save the settings, click "Submit" and then "Apply Config" in the top left corner.
Further Information
Tips and tricks for FreePBX can be found in the FreePBX Tips and Tricks section.