VoIP

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Asterisk

Installing Asterisk in Ubuntu

sudo apt-get install asterisk

Standalone installation:

Download the latest ISO from below link & create a VM from it:

http://www.asterisk.org/downloads/asterisknow
https://wiki.asterisk.org/wiki/display/AST/Installing+AsteriskNOW

LAN Installation

This section will allow you to :

Host calls locally
Access mail boxes

We will be editing the following files:

sip.conf
extensions.conf
voicemail.conf

These files are located in:

/etc/asterisk

Making a backup of the 3 files:

sudo mv /etc/asterisk/sip.conf /etc/asterisk/sip.conf.orig
sudo mv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.orig
sudo mv /etc/asterisk/voicemail.conf /etc/asterisk/voicemail.conf.orig
sip.conf
sudo nano /etc/asterisk/sip.conf
[general]
context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
canreinvite=no
nat=yes
session-timers=refuse
localnet=192.168.1.0/255.255.255.0

[7001]
type=friend
host=dynamic
secret=123
context=internal

[7002]
type=friend
host=dynamic
secret=456
context=internal


extensions.conf
sudo nano /etc/asterisk/extensions.conf
[internal]
exten => 7001,1,Answer()
exten => 7001,2,Dial(SIP/7001,60)
exten => 7001,3,Playback(vm-nobodyavail)
exten => 7001,4,VoiceMail(7001@main)
exten => 7001,5,Hangup()

exten => 7002,1,Answer()
exten => 7002,2,Dial(SIP/7002,60)
exten => 7002,3,Playback(vm-nobodyavail)
exten => 7002,4,VoiceMail(7002@main)
exten => 7002,5,Hangup()

exten => 8001,1,VoicemailMain(7001@main)
exten => 8001,2,Hangup()

exten => 8002,1,VoicemailMain(7002@main)
exten => 8002,2,Hangup()


voicemail.conf
sudo nano /etc/asterisk/voicemail.conf
[main]
7001 => 123

7002 => 456


Reloading Asterisk to apply the configuration

sudo asterisk -rx reload

OR

sudo /etc/init.d/asterisk restart


Conference Call

Source wiki.asterisk.org

Backup original file:

sudo mv /etc/asterisk/confbridge.conf /etc/asterisk/confbridge.conf.orig

Create & Edit the new file:

sudo nano /etc/asterisk/confbridge.conf
[general]
; comments are preceded by a comma
;
; the general section is blank
;
[default_bridge]
type=bridge
; Bridge Profile options go here
myoption=value
myoption2=othervalue
;
[default_user]
type=user
; User Profile options go here
myoption=value
myoption2=othervalue
;
[sample_menu]
type=menu
; Conferece Menu options go here
DTMF=function
otherDTMF=otherFunction
;


[fancybridge]
type=bridge
max_members=20
mixing_interval=10
internal_sample_rate=auto
record_conference=yes


[fancyuser]
type=user
music_on_hold_when_empty=yes
music_on_hold_class=default
announce_user_count_all=yes
announce_join_leave=yes
dsp_drop_silence=yes
denoise=yes
pin=456


[fancymenu]
type=menu
*=playback_and_continue(conf-togglemute&press&digits/1&silence/1&conf-leave&press&digits/2&silence/1&add-a-caller&press&digits/3&silence/1&conf-decrease-talking&press&digits/4&silence/1&reset-talking&press&digits/5&silence/1&increase-talking&press&digits/6&silence/1&conf-decrease-listening&press&digits/7&silence/1&conf-reset-listening&press&digits/8&silence/1&conf-increase-listening&press&digits/9&silence/1&conf-exit-menu&press&digits/0)
*1=toggle_mute
1=toggle_mute
*2=leave_conference
2=leave_conference
*3=dialplan_exec(addcallers,1,1)
3=dialplan_exec(addcallers,1,1)
*4=decrease_listening_volume
4=decrease_listening_volume
*5=reset_listening_volume
5=reset_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=reset_talking_volume
8=reset_talking_volume
*9=increase_talking_volume
9=increase_talking_volume
*0=no_op
0=no_op

[addcaller]
exten => 1,1,Originate(SIP/otherpeer,exten,conferences,100,1)

[conferences]
exten => 100,1,ConfBridge(1234)

Restart asterisk to load new files.

PSTN Calling

Source: howtoforge.com

   Paste.png     This section is under construction.

Addons

   Paste.png     This section is under construction.


Troubleshooting

Asterisk Console:

sudo asterisk -r

Debug Info:

sudo asterisk -rvvv
sudo asterisk -rvvvvvv


References





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