Note
for home users:
Config files manual tweaks in this guide are meant for advanced users. Unless
you have a compelling reason to do so, users are advised against any manual
config file tweaks.
Just as you think that all is OK, you realized something else requires attention. This is true with TRIXBOX as well.
To do this you may need to edit some configuration files (.conf) that reside both in the /etc/asterisk directory and /etc directories. Configuration files in the /etc/asterisk are generally editable through Config Edit.
To
start editing the .conf
files you need to log in to TRIXBOX Configuration and Administration and select
the Config Edit selection.
Select
Config Edit (marked with red arrow) and you will see a new screen
with a list of all the .conf files, which can be edited manually.
You
may scroll down the page to find the file that you wish to edit.
A number of .conf files may require editing to get asterisk to work, depending on the individual requirements. In this guide, as a base, we will be looking at features.conf, sip.conf, iax.conf (just in case you want IAX, but not really necessary if you are just running SIP), indications.conf, enum.conf, extensions_custom.conf, features.conf, zaptel.conf and modprobe.conf (the last 2 files live in the /etc directory may need to be edited if you have a FXO card e.g. X100P installed. We will look a them later on under Interfacing Asterisk to PSTN).
[general]
port = 5060
; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ;
Address to bind to (all addresses on machine)
disallow=all
allow=g729
;
I add this in
sip_custom.conf instead
allow=ulaw
allow=alaw
allow=gsm
; the 2 lines below I
added them in sip_custom.conf although
; you can add them in sip.conf
defaultexpirey=600 ;
**Engin users: include this if necessary
maxexpirey=3600
;
**Engin & BBP Global users: include this if necessary
qualify=yes
progressinband=yes
; If you
need to answer unauthenticated calls, you should
; change this next line to 'from-trunk', rather than
; 'from-sip-external'.
; You'll know this is happening if when you call in you get
; a message saying "The number you have dialed is not in
; service. Please check the number and try again."
context =
from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
; #, in
this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
;;#include additional_a2billing_sip.conf
Note: Every time you update freePBX, there is a good chance that changes or additions you have made in sip.conf will be overwritten. To avoid this, all additions I wish to make in sip.conf are made in sip_custom.conf
language=au
videosupport=yes
allow=g729
allow=g723
allow=h261
allow=h263
allow=h263p
defaultexpirey=600 ; **Engin users: include this if necessary
maxexpirey=3600 ; **Engin & BBP Global users:
include this if necessary
In the
general section, add the following lines if it is not already there:
[general]
delayreject=yes
jitterbuffer=yes
; I placed this line in my iax_custom.conf instead
mailboxdetail=yes
; I placed this line in my iax_custom.conf instead
Note: Every time you update freePBX, there is a good chance that changes or additions you have made in iax.conf will be overwritten. To avoid this, all additions I wish to make in iax.conf are made in iax_custom.conf
; additions to
iax.conf that would not get deleted in you upgrade freepbx
externip = <your fixed external IP> or (one or the other NOT
both)
externhost = <your DynDNS name>
localnet = 192.168.1.0/255.255.255.0
languagae=au
delayreject = yes
jitterbuffer = yes
allow=g729
allow=ilbc
allow=g723
In the general section
ensure the following exist:
[general]
country = au ; (The
default is country = us, therefore replace “us” with “au”)
[general]
;
; The search list for domains may be customized.
; Domains are searched in the order they are listed here.
;
search
=> e164.org
search => e164.arpa
;h323driver => H323 ; this line exists in AAH v2.8
(There
should only be the three entries above).
If
you have a ZAP trunk and you want to get an external line and be presented with
a dial tone first before you dial the external line, you may add the following
codes in the extensions_custom.conf file.
The example below assumes
that your ZAP is on channel 1.
Immediately
after the line following the include => from-internal-trixbox
context, insert the following codes:
exten
=> 0,1,Dial(ZAP/1/{EXTEN})
exten => 0,2,Hangup
This
will allow you to dial 0 and you will get a second dial tone from PSTN. Following which you can dial a PSTN number that you wish to call.
Note:*
If you're using a PRI line, you need to enable 'overlapdial' in /etc/asterisk/zapata.conf
for this to work.
The default features.conf included with trixbox is fairly minimal. It is recommended that you reflect the following in your features.conf file. The extra keywords will come in handy later on.