PBX in a Flash is a lean Asterisk GUI designed to meet the needs of hobbyists as well as business users and VARs. You'll have a high-performance turnkey Asterisk PBX that's easy to upgrade with dozens of add on scripts to provide virtually any feature you can imagine.
And you can choose from tons of Nerd Vittles and FreePBX applications that install in under 15 seconds:
AsteriDex, Weather Reports, News Feeds, Email by Phone, Telephone Reminders, and many more. You add features when you need additional functionality.
If you are not an advanced user of Asterisk, we highly recommend the use of one of GUI interfaces of Asterisk for SIP Configuraiton.
PBX in a Flash as an Astersik GUI is one of the fastest way to get started building custom telephony solutions with Asterisk.
Simply download the .iso file, burn it to a CD, drop it into the CD or DVD drive on the target computer and in less than 30 minutes you will have a full functional Asterisk system ready for your custom telephony application.
PBX in a Flash interface allow administrators to view, edit, and change most aspects of Asterisk via a web interface.
You can download free GUI version of PBX in a Flash 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 PBX in a Flash SIP Configuration.
Below you can find PBX in a Flash 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
