FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, the world's most popular open source telephony engine software. FreePBX has been developed and hardened by thousands of volunteers over tens of thousands man hours.
FreePBX integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface.
If you are not an advanced user of Asterisk, we highly recommend the use of one of GUI interfaces of Asterisk for SIP Configuraiton like FreePBX.
You can download free GUI version of FreePBX from below link:
While our goal is to make all Bring Your Own Device installations as easy as possible, this option is intended for advanced users. VoiceTrunking does not provide technical support for FreePBX SIP Configuration.
Below you can find FreePBX SIP Trunk configuration guide for VoiceTrunking SIP Trunk service.
Outgoing Settings
Peer Details
username=5551231234 (your VoiceTrunking account assigned while signing up)
type=peer
secret=XXXXX (your VoiceTrunking password)
nat=auto
insecure=very
host=sip.VoiceTrunking.com
fromuser=5551231234
(your VoiceTrunking SIP account assigned while signing up)
fromdomain=sip.VoiceTrunking.com
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
Incoming Settings
USER Details
username=5551231234
(your VoiceTrunking SIP account assigned while signing up)
type=user
secret=XXXXXX (your VoiceTrunking password)
nat=auto
insecure=very
host=sip.VoiceTrunking.com
fromdomain=sip.VoiceTrunking.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=g729
allow=ulaw
allow=alaw
allow=ilbc
REGISTRATION STRING
5551231234:XXXXXXXX@sip.voicetrunking.com/55551231234
(for 5551231234 use your VoiceTrunking SIP account and for XXXXXXXX use your VoiceTrunking password)
NOTE: Asterisk does not support DNS SERVER lookups for inbound calls.
If you also have virtual phone number with your SIP Trunk service please add the following line to the "sip_general_custom.conf" file
srvlookup=no
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