See also Build a PBX with YUM Using CentOS, Asterisk and a GUI

See also ASTLinux

http://www.inphonex.com/support/trixbox-configuration-v2.6.1.1.php

If your Asterisk server is behind NAT and you need a remote extension, forward UDP ports 5060, 10000-20000 and 4569 in to the Asterisk server.

Edit sip.conf to include your resolvable external host name and local subnet.

With ASTLinux, it's easiest to use the web editor. Or you can use nano or VI as shown:

vim /etc/asterisk/sip.conf

[general]
externip=external.host.name
localnet=192.168.1.0/255.255.255.0
nat=yes

Note that the first time you log into the GUI via a web browser, your configuration files will be modified! Backup the asterisk configuration directory before logging in:

cp -ar /etc/asterisk /etc/asterisk.backup

You probably want to configure your PBX in the order presented here. It will be easier and make more sense.

Access the web interface using your PBX's hostname or IP address:

http://192.168.1.30:8088/asterisk/static/config/index.html

or try:

http://192.168.1.30:8088/static/config/index.html

Save on each page, then Apply when all done.

• Add analog or VoIP trunks
• Check Status Page for successful registration

VoIP trunks usually require 1+area on all calls.

This is where you define acceptable dial patterns and substitutions.

Dial Plans are groupings of Outgoing Calling Rules. Often the default is sufficient.

CLID specified here will override default (Global) or trunk settings.

You can also edit features.conf for more functionality:

pickupexten = **  ; Configure the pickup extension

Record a greeting (recording) for your IVR.

http://www.asterisktutorials.com/trixbox-ce-creating-an-inbound-dialplan-ivr-menu/

Don't add extensions to filenames in IVRs or Asterisk won't find the files.

exten=i,1,Playback(invalid)
exten=i,2,Goto(s,4)		;go to step 4 of this same IVR

Create rules for specific DIDs plus two 'catchall' rules.

http://kb.digium.com/entry/893/

Backup pbx.js before modification:

cp -a /var/lib/asterisk/static-http/config/js/pbx.js /var/lib/asterisk/static-http/config/js/pbx.js.bak

vi /var/lib/asterisk/static-http/config/js/pbx.js

Use Voice Menus to create and access your custom features.

Is analogous to this custom app:

exten => 65,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 65,n,Echo                     ; Do the echo test
exten => 65,n,Playback(demo-echodone)  ; Let them know it's over
exten => 65,n,Hangup

Use this to test MoH:

Use this for DISA:

Below Not Complete

#include extensions-custom.conf

vim /etc/asterisk/extensions-custom.conf

[DLPN_DialPlan1]+  ; Choose a dialplan context to extend using the plus sign
;
; Create an extension, 66, for evaluating echo latency.
;
exten => 66,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 66,n,Echo                     ; Do the echo [[test]]
exten => 66,n,Playback(demo-echodone)  ; Let them know it's over
exten => 66,n,Hangup
;

Hmm. It seems that the Asterisk-GUI will incorporate the included file into extensions.conf, then delete the #include line. Then, if you modify and save DialPlan1 (in this case), the changes are wiped out completely.

Polycom only, AFAICT.

http://www.savelono.com/linux/how-to-setup-automatic-polycom-provisioning-for-the-asterisk-20-gui.html

Backup hangs:

Check permissions on /var/lib/asterisk/gui-backups. The mini-httpd server runs as nobody on AstLinux…