Asterisk™: The Definitive Guide

x

$ mkdir -p ~/src/asterisk-complete/asterisk
$ cd ~/src/asterisk-complete/asterisk
$ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
$ svn co http://svn.asterisk.org/svn/asterisk/tags/1.8.1
$ cd ~/src/asterisk-complete/asterisk/1.8/
$ ./configure
$ make
$ sudo make install
$ sudo make config
$ cd ~/src/asterisk-complete/asterisk/1.8/
$ make menuselect
$ sudo make install
$ sudo chown -R asteriskpbx:asteriskpbx /usr/lib/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/lib/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/spool/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/log/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/run/asterisk/
$ sudo chown asteriskpbx:asteriskpbx /usr/sbin/asterisk
$ sudo vim /etc/selinux/config
$ sudo mkdir -p /etc/asterisk
$ sudo chown asteriskpbx:asteriskpbx /etc/asterisk
$ cd /etc/asterisk/
$ cp ~/src/asterisk-complete/asterisk/1.8/configs/indications.conf.sample \
./indications.conf
$ cp ~/src/asterisk-complete/asterisk/1.8/configs/asterisk.conf.sample \
/etc/asterisk/asterisk.conf
$ vim /etc/asterisk/asterisk.conf
$ cat >> /etc/asterisk/modules.conf

; The modules.conf file, used to define which modules Asterisk should load (or 
; not load).
;
[modules]
autoload=yes

; Resource modules currently not needed
noload => res_speech.so
noload => res_phoneprov.so
noload => res_ael_share.so
noload => res_clialiases.so
noload => res_adsi.so

; PBX modules currently not needed
noload => pbx_ael.so
noload => pbx_dundi.so

; Channel modules currently not needed
noload => chan_oss.so
noload => chan_mgcp.so
noload => chan_skinny.so
noload => chan_phone.so
noload => chan_agent.so
noload => chan_unistim.so
noload => chan_alsa.so

; Application modules currently not needed
noload => app_nbscat.so
noload => app_amd.so
noload => app_minivm.so
noload => app_zapateller.so
noload => app_ices.so
noload => app_sendtext.so
noload => app_speech_utils.so
noload => app_mp3.so
noload => app_flash.so
noload => app_getcpeid.so
noload => app_setcallerid.so
noload => app_adsiprog.so
noload => app_forkcdr.so
noload => app_sms.so
noload => app_morsecode.so
noload => app_followme.so
noload => app_url.so
noload => app_alarmreceiver.so
noload => app_disa.so
noload => app_dahdiras.so
noload => app_senddtmf.so
noload => app_sayunixtime.so
noload => app_test.so
noload => app_externalivr.so
noload => app_image.so
noload => app_dictate.so
noload => app_festival.so

Ctrl+D
$ cat >> musiconhold.conf

; musiconhold.conf
[default]
mode=files
directory=moh

Ctrl+D
# yum update
Is this ok [y/N]: y
# reboot
# yum remove *.i386 -y
# yum install ntp
...
Is this ok [y/N]: y
...
# ntpdate pool.ntp.org
# chkconfig ntpd on
# service ntpd start
# adduser asteriskpbx
# passwd asteriskpbx
Changing password for user asteriskpbx.
New UNIX password:
Retype new UNIX password:
# yum install sudo
# visudo

## Allows people in group wheel to run all commands
%wheel  ALL=(ALL)       ALL
wheel:x:10:root,asteriskpbx
$ sudo ls /root/
[sudo] password for asteriskpbx:
$ vim ~/.bash_profile
PATH=$PATH:$HOME/bin:/usr/sbin:/sbin
$ sudo apt-get update
[sudo] password for asteriskpbx:
...
Reading package lists... Done

$ sudo apt-get upgrade
Reading state information... Done
...
Do you want to continue [Y/n]? y
$ sudo reboot
$ sudo apt-get install ntp
$ sudo nano /etc/ntp.conf
# By default, exchange time with everybody, but don't allow configuration.
restrict -4 default kod notrap nomodify nopeer noquery
restrict -6 default kod notrap nomodify nopeer noquery
# By default, exchange time with everybody, but don't allow configuration.
restrict -4 default kod notrap nomodify nopeer noquery
restrict -6 default kod notrap nomodify nopeer noquery

restrict -4 127.0.0.1
restrict -6 ::1
$ sudo /etc/init.d/ntp restart
$ cd ~/src/asterisk-complete/asterisk/1.8
$ sudo ./contrib/scripts/install_prereq install
$ sudo ./contrib/scripts/install_prereq install-unpackaged
$ mkdir -p ~/src/asterisk-complete/asterisk
$ cd ~/src/asterisk-complete/asterisk
$ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
$ cd ~/src/asterisk-complete/asterisk
$ wget \
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz
$ tar zxvf asterisk-1.8-current.tar.gz
$ cd ~/src/asterisk-complete/
$ mkdir libpri
$ cd libpri/
$ svn co http://svn.asterisk.org/svn/libpri/tags/1.4.<your version number>
$ cd 1.4.<your version number>
$ make
$ sudo make install
$ cd ~/src/asterisk-complete/
$ mkdir dahdi
$ cd dahdi/
$ svn co http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.4.0+2.4.0
$ cd 2.4.0+2.4.0
$ make
$ sudo make install
$ sudo make config
$ cd ~/src/asterisk-complete/asterisk/1.8
$ ./configure
$ make
$ sudo make install
$ sudo make config
$ sudo chown -R asteriskpbx:asteriskpbx /usr/lib/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/lib/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/spool/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/log/asterisk/
$ sudo chown -R asteriskpbx:asteriskpbx /var/run/asterisk
$ sudo chown asteriskpbx:asteriskpbx /usr/sbin/asterisk
SUBSYSTEM=="dahdi", OWNER="asteriskpbx", GROUP="asteriskpbx", MODE="0660"
$ sudo echo 0 > /selinux/enforce
$ sudo echo 1 > /selinux/enforce
$ cd /etc/selinux/
$ sudo vim config
# This file controls the state of SELinux on the system.
# SELINUX= can take one of these three values:
#       enforcing - SELinux security policy is enforced.
#       permissive - SELinux prints warnings instead of enforcing.
#       disabled - SELinux is fully disabled.
SELINUX=disabled
# SELINUXTYPE= type of policy in use. Possible values are:
#       targeted - Only targeted network daemons are protected.
#       strict - Full SELinux protection.
SELINUXTYPE=targeted

# SETLOCALDEFS= Check local definition changes
SETLOCALDEFS=0
$ sudo reboot
$ sudo mkdir /etc/asterisk/
$ sudo chown asteriskpbx:asteriskpbx /etc/asterisk/
$ cp ~/src/asterisk-complete/asterisk/1.8/configs/indications.conf.sample \
/etc/asterisk/indications.conf
$ cp ~/src/asterisk-complete/asterisk/1.8/configs/asterisk.conf.sample \
/etc/asterisk/asterisk.conf
$ vim /etc/asterisk/asterisk.conf
runuser=asteriskpbx
rungroup=asteriskpbx
$ /usr/sbin/asterisk -cvvv
*CLI> module show
Module                         Description                              Use Count 
0 modules loaded
*CLI> core stop now
$ cat >> /etc/asterisk/modules.conf
          ; The modules.conf file, used to define which modules Asterisk should load (or
; not load).
;
[modules]
autoload=yes

$ asterisk -c
*CLI> module show
Module                         Description                              Use Count 
res_speech.so                  Generic Speech Recognition API           0
res_monitor.so                 Call Monitoring Resource                 0
...
func_math.so                   Mathematical dialplan function           0         
171 modules loaded
; Resource modules
noload => res_speech.so
noload => res_phoneprov.so
noload => res_ael_share.so
noload => res_clialiases.so
noload => res_adsi.so

; PBX modules
noload => pbx_ael.so
noload => pbx_dundi.so

; Channel modules
noload => chan_oss.so
noload => chan_mgcp.so
noload => chan_skinny.so
noload => chan_phone.so
noload => chan_agent.so
noload => chan_unistim.so
noload => chan_alsa.so

; Application modules
noload => app_nbscat.so
noload => app_amd.so
noload => app_minivm.so
noload => app_zapateller.so
noload => app_ices.so
noload => app_sendtext.so
noload => app_speech_utils.so
noload => app_mp3.so
noload => app_flash.so
noload => app_getcpeid.so
noload => app_setcallerid.so
noload => app_adsiprog.so
noload => app_forkcdr.so
noload => app_sms.so
noload => app_morsecode.so
noload => app_followme.so
noload => app_url.so
noload => app_alarmreceiver.so
noload => app_disa.so
noload => app_dahdiras.so
noload => app_senddtmf.so
noload => app_sayunixtime.so
noload => app_test.so
noload => app_externalivr.so
noload => app_image.so
noload => app_dictate.so
noload => app_festival.so
$ cd /etc/asterisk/
$ cat >> musiconhold.conf
; musiconhold.conf
[default]
mode=files
directory=moh

Ctrl+D
$ cd ~/src/asterisk-complete/asterisk/1.8.<your version>/
$ cd menuselect
$ make clean
$ ./configure
$ cd ..
$ make menuselect
$ cd ~/src/asterisk-complete/asterisk/1.8.<your version>/
$ make menuselect
$ sudo make install
$ sudo chown -R asteriskpbx:asteriskpbx /var/lib/asterisk/sounds/
$ cd ~/src/asterisk-complete/asterisk/1.8.<your version>/
$ ./configure
$ cd menuselect
$ make menuselect
$ cd ..
$ make menuselect-tree
Generating input for menuselect ...
Usage: menuselect/menuselect [--enable <option>] [--disable <option>]
   [--enable-category <category>] [--enable-all]
   [--disable-category <category>] [--disable-all] [...]
   [<config-file> [...]]
Usage: menuselect/menuselect { --check-deps | --list-options
   | --list-category <category> | --category-list | --help }
   [<config-file> [...]]
$ menuselect/menuselect --disable-all menuselect.makeopts
$ menuselect/menuselect --disable-all --enable chan_sip \
--enable app_dial menuselect.makeopts
$ cd ~/src/asterisk-complete/asterisk/1.8.<your version number>/
$ ./configure
$ make
$ make install
 WARNING WARNING WARNING

 Your Asterisk modules directory, located at
 /usr/lib/asterisk/modules
 contains modules that were not installed by this
 version of Asterisk. Please ensure that these
 modules are compatible with this version before
 attempting to run Asterisk.

    chan_mgcp.so
    chan_oss.so
    chan_phone.so
    chan_skinny.so
    chan_skype.so
    codec_g729a.so
    res_skypeforasterisk.so

 WARNING WARNING WARNING
$ cd ~/src/asterisk-complete/
$ mkdir thirdparty/
$ mkdir thirdparty/1.8/
$ sudo asterisk -C /custom/path/to/asterisk.conf
$ man asterisk
Set(CHANNEL(tonezone)=[yourcountry]) ; i.e., uk, de, etc.
[starwars](us)
description = Star Wars Theme Song
ring = 262/400,392/500,0/100,349/400,330/400,294/400,524/400,392/500,0/100,349/400, \
       330/400,294/400,524/400,392/500,0/100,349/400,330/400,349/400,294/500,0/2000

exten => 500,1,Answer()
    same => n,Set(CHANNEL(tonezone)=starwars)
    same => n,Dial(SIP/0000FFFF0002) ; or whatever your channel is named in sip.conf

$ rpm -Uhv http://apt.sw.be/redhat ...
$ yum install yum-priorities
$ yum install mpg123
$ mpg123 -w SilentCity.wav SilentCity.mp3
$ sox SilentCity.wav -t raw -r 8000 -s -w -c 1 SilentCity.sln
# sudo apt-get install sox libsox-fmt-all
$ sox SilentCity.mp3 -t raw -r 8000 -s -w -c 1 SilentCity.sln
$ cp *.sln /var/lib/asterisk/moh
$ asterisk -rx "module unload res_musiconhold.so"
$ asterisk -rx "module load res_musiconhold.so"
exten => 664,1,NoOp()
    same => n,Progress()
    same => n,MusicOnHold()

[general]
context=unauthenticated         ; default context for incoming calls
allowguest=no                   ; disable unauthenticated calls
srvlookup=yes                   ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0             ; listen for UDP requests on all interfaces
tcpenable=no                    ; disable TCP support

[office-phone](!)   ; create a template for our devices
type=friend         ; the channel driver will match on username first, IP second
context=LocalSets   ; this is where calls from the device will enter the dialplan
host=dynamic        ; the device will register with asterisk
nat=yes             ; assume device is behind NAT
                    ; *** NAT stands for Network Address Translation, which allows  
                    ; multiple internal devices to share an external IP address.
secret=s3CuR#p@s5   ; a secure password for this device -- DON'T USE THIS PASSWORD!
dtmfmode=auto       ; accept touch-tones from the devices, negotiated automatically
disallow=all        ; reset which voice codecs this device will accept or offer
allow=ulaw          ; which audio codecs to accept from, and request to, the device
allow=alaw          ; in the order we prefer

; define a device name and use the office-phone template
[0000FFFF0001](office-phone)

; define another device name using the same template
[0000FFFF0002](office-phone)
          $ 
          dd if=/dev/random count=1 bs=8 2>/dev/null | base64 | sed -e 's/=*$//'

[0000FFFF0003](office-phone) ; template must be on same line and no space between
secret=@N0th3rP4S5
allow=gsm
[0000FFFF0003]

;
; These options came from the template section.
;
type=friend
context=LocalSets
host=dynamic
nat=yes 
secret=s3CuR#p@s5
dtmfmode=auto
disallow=all
allow=ulaw
allow=alaw

;
; These options came from the specific section.
;
secret=@N0th3rP4S5 ; Overrides the secret from the template
allow=gsm          ; Adds gsm to the list of allowed codecs
[general]
autokill=yes   ; don't stall for a long time if other endpoint doesn't respond
srvlookup=yes  ; enable DNS SRV lookups for outbound calls

[office-phone](!)    ; template for IAX-based office phones
type=friend          ; Asterisk will allow calls to and from this phone
host=dynamic         ; inform Asterisk that the phone will tell us where
                     ;   it is on the network
secret=my5UP3rp@s5!  ; a secure password -- which means DON'T USE THIS ONE!
context=LocalSets    ; the context where incoming requests will enter the dialplan
disallow=all         ; reset the available voice codecs
allow=ulaw           ; prefer the ulaw codec
allow=alaw           ; but also allow the alaw codec

[0000FFFF0004](office-phone) ; define our first phone with the office-phone template

[0000FFFF0005](office-phone) ; define another phone with our office-phone template
$ dd if=/dev/random count=1 bs=8 2>/dev/null | base64 | sed -e 's/=*$//'
[0004F2FF0006](office-phone)
disallow=all ; Clear previous codec settings.
allow=gsm    ; Allow only gsm.
[0004f2119698](office-phone)

[001b63a28ccc](office-phone) ; you could have a comment like Leif's Laptop

$ sudo asterisk -r
*CLI> module reload chan_sip.so
*CLI> module reload chan_iax2.so
*CLI> sip show peers
*CLI> sip show users
*CLI> iax2 show peers
*CLI> iax2 show users
$ sudo asterisk -r
*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
0000FFFF0001/0000FFFF0001    192.168.1.100    D   N      5060     Unmonitored
0000FFFF0002/0000FFFF0002    192.168.1.101    D   N      5060     Unmonitored

# Autogenerated by tools/xpp/dahdi_genconf (Dahdi::Config::Gen::Modules) on 
# Tue Jul 27 10:31:46 2010
# If you edit this file and execute tools/xpp/dahdi_genconf again,
# your manual changes will be LOST.
wctdm24xxp

# Specify that we would like DAHDI to generate tones that are
# used in the United States.
loadzone = us
defaultzone = us

# We have 4 FXS ports; configure them to use FXO signaling.
fxoks = 1-4
echocanceller = mg2,1-4
$ sudo /etc/init.d/dahdi start
Loading DAHDI hardware modules:
  wctdm24xxp:                                              [  OK  ]

Running dahdi_cfg:                                         [  OK  ]
*CLI> module show like chan_dahdi.so
Module                         Description                              Use Count 
chan_dahdi.so                  DAHDI Telephony Driver                   0         
1 modules loaded

[trunkgroups]

; No trunk groups are needed in this configuration.

[channels]

; The channels context is used when defining channels using the
; older deprecated method.  Don't use this as a section name.

[phone](!)
;
; A template to hold common options for all phones.
;
usecallerid = yes
hidecallerid = no
callwaiting = no
threewaycalling = yes
transfer = yes
echocancel = yes
echotraining = yes
immediate = no
context = LocalSets
signalling = fxo_ks ; Uses FXO signaling for an FXS channel 

[phone1](phone)
callerid = "Mark Michelson" <(256)555-1212>
dahdichan = 1

[phone2](phone)
callerid = "David Vossel" <(256)555-2121>
dahdichan = 2

[phone3](phone)
callerid = "Jason Parker" <(256)555-3434>
dahdichan = 3

[phone4](phone) 
callerid = "Matthew Nicholson" <(256)555-4343>
dahdichan = 4

*CLI> dahdi show channels
   Chan Extension  Context         Language   MOH Interpret    Blocked    State     
 pseudo            default                    default                     In Service
      1            LocalSets                  default                     In Service
      2            LocalSets                  default                     In Service
      3            LocalSets                  default                     In Service
      4            LocalSets                  default                     In Service
[LocalSets]

exten => 100,1,Dial(SIP/0000FFFF0001) ; Replace 0000FFFF0001 with your device name

exten => 101,1,Dial(SIP/0000FFFF0002) ; Replace 0000FFFF0002 with your device name

;
; These will allow you to dial each of the 4 analog phones configured
; in the previous section.
;
exten => 102,1,Dial(DAHDI/1)
exten => 103,1,Dial(DAHDI/2)
exten => 104,1,Dial(DAHDI/3)
exten => 105,1,Dial(DAHDI/4)

exten => 200,1,Answer()
    same => n,Playback(hello-world)
    same => n,Hangup()

$ sudo asterisk -rx "dialplan reload"
*CLI> dialplan reload
    -- Executing [100@LocalSets:1] Dial("SIP/0000FFFF0001-0000000c", 
       "SIP/0000FFFF0001") in new stack
    -- Called 0000FFFF0001
    -- SIP/0000FFFF0001-0000000d is ringing
[incoming]
exten =>
exten => name,priority,application()
exten => 123,1,Answer()
exten => 123,1,Answer()
exten => 123,2,Hangup()
exten => 123,1,Answer()
exten => 123,2,do something
exten => 123,3,do something else
exten => 123,4,do one last thing
exten => 123,5,Hangup()
exten => 123,1,Answer()
exten => 123,n,do something
exten => 123,n,do something else
exten => 123,n,do one last thing
exten => 123,n,Hangup()
exten => 123,1,Answer()
   same => n,do something
   same => n,do something else
   same => n,do one last thing
   same => n,Hangup()
exten => 123,n(label),application()
exten => 123,n,(label),application() ;<-- THIS IS NOT GOING TO WORK
Playback(/home/john/sounds/filename)
Playback(custom/filename)
exten => 200,1,Answer()
   same => n,Playback(hello-world)
   same => n,Hangup()
[LocalSets] ; this is the context name
exten => 100,1,Dial(SIP/0000FFFF0001) ; Replace 0000FFFF0001 with your device name

exten => 101,1,Dial(SIP/0000FFFF0002) ; Replace 0000FFFF0002 with your device name

exten => 200,1,Answer()
   same => n,Playback(hello-world)
   same => n,Hangup()
*CLI> dialplan reload
$ sudo /usr/sbin/asterisk -rx "dialplan reload"
    same => n,Goto(context,extension,priority)
exten => 201,1,Goto(TestMenu,start,1) ; add this to the end of the 
                                      ; [LocalSets] context

[TestMenu]
exten => start,1,Answer()
[TestMenu]
exten => start,1,Answer()
   same => n,Background(main-menu)
[TestMenu]
exten => start,1,Answer()
   same => n,Background(main-menu)
   same => n,WaitExten()
   same => n,WaitExten(5) ; We recommend always passing a time argument to WaitExten()
[TestMenu]
exten => start,1,Answer()
   same => n,Background(main-menu)
   same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)

exten => 2,1,Playback(digits/2)
*CLI> dialplan reload
[TestMenu]
exten => start,1,Answer()
   same => n,Background(main-menu)
   same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
   same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
   same => n,Goto(TestMenu,start,1)

[TestMenu]
exten => start,1,Answer()
   same => n,Background(main-menu)
   same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
   same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
   same => n,Goto(TestMenu,start,1)

exten => i,1,Playback(pbx-invalid)
   same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
   same => n,Hangup()
exten => 105,1,Dial(DAHDI/1)
exten => 105,1,Dial(DAHDI/1&SIP/0004F2001122&IAX2/Softphone)
Dial(technology/user[:password]@remote_host[:port][/remote_extension])
exten => 500,1,Dial(IAX2/guest@misery.digium.com/s)
Dial(DAHDI/[gGrR]channel_or_group[/remote_extension])
exten => 501,1,Dial(DAHDI/4/18005551212)
exten => 201,1,Dial(DAHDI/1,10)
exten => 201,1,Dial(DAHDI/1,10)    
   same => n,Playback(vm-nobodyavail)
   same => n,Hangup()

exten => 201,1,Dial(DAHDI/1,10,m)
   same => n,Playback(vm-nobodyavail)
   same => n,Hangup()
[TestMenu]
exten => start,1,Answer()
   same => n,Background(main-menu)
   same => n,WaitExten(5)

exten => 1,1,Dial(SIP/0000FFFF0001,10) ; Replace 0000FFFF0001 with your device name
   same => n,Playback(vm-nobodyavail)
   same => n,Hangup()

exten => 2,1,Dial(SIP/0000FFFF0002,10) ; Replace 0000FFFF0002 with your device name
   same => n,Playback(vm-nobodyavail)
   same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
   same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
   same => n,Hangup()
exten => 1,1,Dial(DAHDI/1,,m)
exten => 301,1,Set(LEIF=SIP/0000FFFF0001)
   same => n,Dial(${LEIF})
[globals]
LEIF=SIP/0000FFFF0001
exten => 202,1,Set(MagicNumber=42)
    same => n,SayNumber(${MagicNumber})
[globals]
LEIF=SIP/0000FFFF0001
JIM=SIP/0000FFFF0002
RUSSELL=SIP/0000FFFF0003

[LocalSets]
exten => 100,1,Dial(${LEIF})
exten => leif,1,Dial(${LEIF})

exten => 101,1,Dial(${JIM})
exten => jim,1,Dial(${JIM})

exten => 102,1,Dial(${RUSSELL})
exten => russell,1,Dial(${RUSSELL})

[TestMenu]
exten => 201,1,Answer()
   same => n,Background(enter-ext-of-person)
   same => n,WaitExten()

exten => 1,1,Dial(DAHDI/1,10)
   same => n,Playback(vm-nobodyavail)
   same => n,Hangup()

exten => 2,1,Dial(SIP/Jane,10)
   same => n,Playback(vm-nobodyavail)
   same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
   same => n,Goto(incoming,123,1)

exten => t,1,Playback(vm-goodbye)
   same => n,Hangup()
_.
_X.
_[0-9a-zA-Z].
exten => _NXX,1,Playback(silence/1&auth-thankyou)
exten => _555XXXX,1,Playback(silence/1&digits/1)
exten => _55512XX,1,Playback(silence/1&digits/2)
_NXXXXXX
_NXXNXXXXXX
_1NXXNXXXXXX
_011.
; UK, Germany, Italy, China, etc.
_00. ; international dialing code
_0. ; national dialing prefix

; Australia
_0011. ; international dialing code
_0. ; national dialing prefix

exten => _XXX,1,Answer()
   same => n,SayDigits(${EXTEN})
exten => _XXX,1,Answer()
   same => n,SayDigits(${EXTEN:1})
94169671111
include => context
[globals]
LOCAL=DAHDI/G0           ; assuming you have a PSTN card in your system
TOLL=SIP/YourVoipCarrier ; as defined in sip.conf

[external]
exten => _NXXNXXXXXX,1,Dial(${LOCAL}/${EXTEN})  ; 10-digit pattern match for NANP
exten => _NXXXXXX,1,Dial(${LOCAL}/${EXTEN})    ; 7-digit pattern match for NANP
exten => _1NXXNXXXXXX,1,Dial(${TOLL}/${EXTEN}) ; Long-distance pattern match for NANP
exten => _011.,1,Dial(${TOLL}/${EXTEN})        ; International pattern match for 
                                               ; calls made from NANP

; This section is functionally the same as the above section. 
; It is for people who like to dial '9' before their calls
exten => _9NXXNXXXXXX,1,Dial(${LOCAL}/${EXTEN:1})
exten => _9NXXXXXX,1,Dial(${LOCAL}/${EXTEN:1})
exten => _91NXXNXXXXXX,1,Dial(${TOLL}/${EXTEN:1})
exten => _9011.,1,Dial(${TOLL}/${EXTEN:1})

[LocalSets]
include => external
# Contains the list of modules to be loaded / unloaded by /etc/init.d/dahdi. 
# 
# NOTE: Please add/edit /etc/modprobe.d/dahdi or /etc/modprobe.conf if you 
# would like to add any module parameters. 
# 
# Format of this file: list of modules, each in its own line. 
# Anything after a '#' is ignored, likewise trailing and leading 
# whitespace and empty lines. 
# Digium TE205P/TE207P/TE210P/TE212P: PCI dual-port T1/E1/J1 
# Digium TE405P/TE407P/TE410P/TE412P: PCI quad-port T1/E1/J1 
# Digium TE220: PCI-Express dual-port T1/E1/J1 
# Digium TE420: PCI-Express quad-port T1/E1/J1 
#wct4xxp 
# Digium TE120P: PCI single-port T1/E1/J1 
# Digium TE121: PCI-Express single-port T1/E1/J1 
# Digium TE122: PCI single-port T1/E1/J1 
#wcte12xp 
# Digium T100P: PCI single-port T1 
# Digium E100P: PCI single-port E1 
#wct1xxp 
# Digium TE110P: PCI single-port T1/E1/J1 
#wcte11xp 
# Digium TDM2400P/AEX2400: up to 24 analog ports 
# Digium TDM800P/AEX800: up to 8 analog ports 
# Digium TDM410P/AEX410: up to 4 analog ports 
#wctdm24xxp 
# X100P - Single port FXO interface 
# X101P - Single port FXO interface 
#wcfxo 
# Digium TDM400P: up to 4 analog ports 
#wctdm 
# Digium B410P: 4 NT/TE BRI ports 
#wcb4xxp 
# Digium TC400B: G729 / G723 Transcoding Engine 
#wctc4xxp 
# Xorcom Astribank Devices 
#xpp_usb
$ sudo /etc/init.d/dahdi restart 
Unloading DAHDI hardware modules: done 
Loading DAHDI hardware modules: 
No hardware timing source found in /proc/dahdi, loading dahdi_dummy 
Running dahdi_cfg: [ OK ]
loadzone = us
defaultzone = us

span = 1,1,0,esf,b8zs
bchan = 1-23
echocanceller = mg2,1-23
hardhdlc = 24

[trunkgroups]

[channels]

usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
echocancelwhenbridged = yes
relaxdtmf = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no

switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0
echocancel = yes
signalling = pri_cpe
channel => 1-23

switchtype = dms100
span = 1,0,0,ccs,hdb3,crc4
bchan = 1-15,17-31
hardhdlc = 16
[trunkgroups]

[channels]

usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
echocancelwhenbridged = yes
relaxdtmf = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no

switchtype = qsig
context = pri_incoming
group = 0
signalling = pri_cpe
channel => 1-15,17-31

$ r2test -l
Variant Code        Country                                 
AR                  Argentina                               
BR                  Brazil                                  
CN                  China                                   
CZ                  Czech Republic                          
CO                  Colombia                                
EC                  Ecuador                                 
ITU                 International Telecommunication Union   
MX                  Mexico                                  
PH                  Philippines                             
VE                  Venezuela
$ ls doc/asterisk/
ar  br  ec  mx  ve

loadzone = us
defaultzone = us

span = 1,1,0,cas,hdb3
cas = 1-15:1101
cas = 17-31:1101

span = 2,1,0,cas,hdb3
cas = 32-46:1101
cas = 48-62:1101

signalling = mfcr2
mfcr2_variant = mx
mfcr2_get_ani_first = no
mfcr2_max_ani = 10
mfcr2_max_dnis = 4
mfcr2_category = national_subscriber
mfcr2_mfback_timeout = -1
mfcr2_metering_pulse_timeout = -1
; this is for debugging purposes
mfcr2_logdir = log
mfcr2_logging = all
; end debugging configuration
channel => 1-15
channel => 17-31

loadzone = us       ; tonezone defines sounds the interface must produce
                    ; (dialtone, busy signal, ringback, etc.)
defaultzone = us    ; define a default tonezone
fxsks = 1-4         ; which channels on the card will have these parameters

[channels]

;
; To apply other options to these channels, put them before "channel".
;
signalling = fxs_ks  ; in Asterisk, FXO channels use FXS signaling 
                     ; (and yes, FXS channels use FXO signaling)
channel => 1-4       ; apply all the previously defined settings to this channel

[incoming]
exten => s,1,Answer()
   same => n,Playback(tt-weasels)
   same => n,Hangup()
[from-voip-network]
exten => _X.,1,Verbose(2, Call from VoIP network to ${EXTEN})
   same => n,Dial(DAHDI/g0/${EXTEN})
[from-pstn] 
; This is the context that would be listed in the config file
; for the circuit (i.e. chan_dahdi.conf)

exten => _X.,1,Verbose(2,Incoming call to ${EXTEN})
    same => n,Goto(number-mapping,${EXTEN},1)

[number-mapping] 
; This context is not strictly required, but will make it easier
; to keep track of your DIDs in a single location in your dialplan. 
; From here you can pass the call to another part of the dialplan
; where the actual dialplan work will take place.

exten => 4165551234,1,Dial(SIP/0000FFFF0001)
exten => 4165554321,1,Goto(autoattendant-context,start,1)
exten => 4165559876,1,VoiceMailMain() ; a handy back door for listening
                                      ; to voice messages

exten => i,1,Verbose(2,Incoming call to invalid number)

[serverB]

;
; Specify the SIP account type as 'peer'.  This means that incoming
; calls will be matched on IP address and port number.  So, when Asterisk
; receives a call from 192.168.1.102 and the standard SIP port of 5060,
; it will match this entry in sip.conf.  It will then request authentication
; and expect the password to match the 'secret' specified here.
;
type = peer
;
; This is the IP address for the remote box (serverB). This option can also
; be provided a hostname.
;
host = 192.168.1.102
;
; When we send calls to this SIP peer and must provide authentication,
; we use 'serverA' as our username.
;
username = serverA
;
; This is the shared secret with serverB.  It will be used as the password
; when either receiving a call from serverB, or sending a call to serverB.
;
secret = apples
;
; When receiving a call from serverB, match it against extensions
; in the 'incoming' context of extensions.conf.
;
context = incoming
;
; Start by clearing out the list of allowed codecs.
;
disallow = all
;
; Only allow the ulaw codec.
;
allow = ulaw

[serverA]

type = peer
host = 192.168.1.101
username = serverB
secret = apples
context = incoming
disallow = all
allow = ulaw

*CLI> sip show peers
Name/username       Host             Dyn Forcerport ACL Port     Status     
serverB/serverA     192.168.1.101                       5060     Unmonitored 
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]
exten => _6XXX,1,Dial(SIP/${EXTEN}@serverB)
[general]
...
register => username:password@your.provider.tld
...
[myprovider]

type = peer
host = your.provider.tld
username = username
secret = password
; Most providers won't authenticate when they send calls to you,
; so you need this line to just accept their calls.
insecure = invite
dtmfmode = rfc2833
disallow = all
allow = ulaw
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider)
[general]

tlsenable = yes
tlsbindaddr = ::

$ cd contrib/scripts
$ mkdir certs
$ ./ast_tls_cert -d certs -C serverA -o serverA
$ ./ast_tls_cert -d certs -C serverB -o serverB -c certs/ca.crt -k certs/ca.key
$ ls certs
ca.cfg  ca.crt  ca.key  serverA.crt  serverA.csr  serverA.key  serverA.pem
serverB.crt  serverB.csr  serverB.key  serverB.pem  tmp.cfg

[general]

tlscertfile = /var/lib/asterisk/keys/serverA.pem
tlscafile = /var/lib/asterisk/keys/ca.crt
[general]

tlscertfile = /var/lib/asterisk/keys/serverB.pem
tlscafile = /var/lib/asterisk/keys/ca.crt
[serverB]

type = peer
host = 192.168.1.102
username = serverA
secret = apples
context = incoming                                                           
disallow = all                    
allow = ulaw
transport = tls                   

[serverA]

type = peer
host = 192.168.1.101
username = serverB
secret = apples
context = incoming                                                           
disallow = all                    
allow = ulaw
transport = tls   
[default]

exten => 1234,1,Set(CHANNEL(secure_bridge_signaling)=1)
    same => n,Dial(SIP/1234@serverB)

[incoming]

exten => _X.,1,Answer()
    same => n,GotoIf($["${CHANNEL(secure_signaling)}" = "1"]?secure:insecure)
    same => n(secure),NoOp(Signaling is encrypted.)
    same => n,Hangup()
    same => n(insecure),NoOp(Signaling is not encrypted.)
    same => n,Hangup()

    -- Executing [1234@incoming:1] Answer("SIP/serverA-00000000", "") in new stack
    -- Executing [1234@incoming:2] GotoIf("SIP/serverA-00000000", 
       "1?secure:insecure") in new stack
    -- Goto (incoming,1234,3)
    -- Executing [1234@incoming:3] NoOp("SIP/serverA-00000000", 
       "Signaling is encrypted.") in new stack
    -- Executing [1234@incoming:4] Hangup("SIP/serverA-00000000", "") in new stack

*CLI> module show like res_srtp.so 
Module                         Description                              Use Count 
res_srtp.so                    Secure RTP (SRTP)                        0         
1 modules loaded

[default]

exten => 1234,1,Set(CHANNEL(secure_bridge_signaling)=1)
    same => n,Set(CHANNEL(secure_bridge_media)=1)
    same => n,Dial(SIP/1234@serverB)

[general]
encryption = yes

forceencryption = yes
[emergency-services]
exten => 911,1,Goto(dialpsap,1)
exten => 9911,1,Goto(dialpsap,1) ; some people will dial '9' because 
                                 ; they're used to doing that from the PBX
exten => 999,1,Goto(dialpsap,1)
exten => 112,1,Goto(dialpsap,1)

exten => dialpsap,1,Verbose(1,Call initiated to PSAP!)
    same => n,Dial(${LOCAL}/911) ; REPLACE 911 HERE WITH WHATEVER 
                                 ; IS APPROPRIATE TO YOUR AREA

[internal]
include => emergency-services   ; you have to have this in any context 
                                ; that has users in it

[no-emergency-services]
exten => 911,1,Goto(nopsap,1)
exten => 9911,1,Goto(nopsap,1) ; for people who dial '9' before external calls
exten => 999,1,Goto(nopsap,1)
exten => 112,1,Goto(nopsap,1)

exten => nopsap,1,Verbose(1,Call initiated to PSAP!) 
    same => n,Playback(no-emerg-service) ; you'll need to record this prompt

[remote-users]
include => no-emergency-services
mailbox context oldpass newpass
mailbox => password[,FirstName LastName[,email addr[,pager addr[,options[|options]]]]]
[default]
100 => 5542,Mike Loukides,mike@shifteight.org
101 => 67674,Tim OReilly,tim@shifteight.org
102 => 36217,Mary JonesSmith,mary.jones-smith@shifteight.org

; *** This needs to all be on the same line
103 => 5426,Some Guy,,,dialout=fromvm|callback=fromvm
|review=yes|operator=yes|envelope=yes

[shifteight]
100 => 0107,Leif Madsen,leif@shifteight.org
101 => 0523,Jim VanMeggelen,jim@shifteight.org,,attach=no|maxmsg=100
102 => 11042,Tilghman Lesher,,,attach=no|tz=central
; Voicemail Configuration

[general]
format=wav49|wav
serveremail=voicemail@shifteight.org
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
pagerdateformat=%A, %B %d, %Y at %r
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM

[shifteight.org]
100 => 1234,Leif Madsen,leif@shifteight.org
101 => 1234,Jim Van Meggelen,jim@shifteight.org
102 => 1234,Russell Bryant,russell@shifteight.org
103 => 1234,Jared Smith,jared@shifteight.org

VoiceMail(mailbox[@context][&mailbox[@context][&...]][,options])
exten => 101,1,Dial(${JOHN})
exten => 101,1,Dial(${JOHN},10)
exten => 101,n,VoiceMail(101@default,u)      
exten => 101,1,Dial(${JOHN},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(101@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(101@default,b)
   same => n,Hangup()
VoiceMailMain([mailbox][@context][,options])
[Services]

exten => *98,1,VoiceMailMain()
exten => 8,1,Directory(default,incoming,f)
exten => 9,1,Directory(default,incoming)
[Services]

exten => *98,1,Dial(Local/vmm@Services/nj)

exten => vmm,1,VoiceMailMain()
/var/spool/asterisk/voicemail/default
./INBOX
./Old
./Old/msg0000.WAV
./Old/msg0000.txt
./Old/msg0001.WAV
./Old/msg0001.txt
./Urgent
./busy.WAV
./unavail
./unavail.WAV

[messagewaiting](!)            ; a template to handle the settings common 
                               ; to all mailboxes
type=peer
subscribecontext=voicemailbox  ; the dialplan context on the voicemail server
context=voicemailbox           ; the dialplan context on the voicemail server
host=192.168.1.1               ; ip address of presence server

[0000FFFF0001](messagewaiting) ; this will need to match the subscriber 
                               ; name on the proxy
mailbox=0000FFFF0001@DIR1      ; this has to be in the form mailbox@mailboxcontext
defaultuser=0000FFFF0001       ; this will need to match the subscriber 
                               ; name on the proxy

exten => sendtovoicemail,1,Verbose(2,Set SIP headers for voicemail)
    same => n,SipAddHeader(X-Voicemail-Mailbox: <mailbox number>)
    same => n,SipAddHeader(X-Voicemail-Context: voicemailbox)
[voicemailbox]
; direct incoming calls to a mailbox
exten =>  Deliver,1,NoOp()
    same =>  n,Set(Mailbox=${SIP_HEADER(X-Voicemail-Mailbox)})
    same =>  n,Set(MailboxContext=${SIP_HEADER(X-Voicemail-Context)})
    same =>  n,VoiceMail(${Mailbox}@${MailboxContext})
    same =>  n,Hangup()

; connect users to their mailbox so that they can retrieve messages exten =>  
Retrieve,1,NoOp()
    same =>  n,Set(Mailbox=${SIP_HEADER(X-Voicemail-Mailbox)})
    same =>  n,Set(MailboxContext=${SIP_HEADER(X-Voicemail-Context)})
    same =>  n,VoiceMailMain(${Mailbox}@${MailboxContext})
    same =>  n,Hangup()
[VOICEMAILTRUNK]
type=peer
defaultuser=voicemail
fromuser=voicemail
secret=s0m3th1ngs3cur3
canreinvite=no
host=<address of proxy/registrar server>
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=voicemailbox
smdienable=yes
smdiport=/dev/ttyS0; or whatever serial port you are connecting your SMDI service to
[interfaces]
charsize=7
paritybit=even
baudrate=1200           ; hopefully a higher bitrate is supported
smdiport=/dev/ttyS0     ; or whatever serial port you'll be using to handle 
                        ; SMDI messages on asterisk

[mailboxes]             ; map incoming digit strings (typically DID numbers) 
                        ; to a valid mailbox@context in voicemail.conf
smdiport=/dev/ttyS0     ; first declare which SMDI port the following mailboxes 
                        ; will use
4169671111=1234@default
4165551212=9999@default

SMDI_MSG_RETRIEVE(<smdi port>,<search key>[,timeout[,options]])
SMDI_MSG(<message_id>,<component>)
# Tone Zone Data
# ^^^^^^^^^^^^^^
# Finally, you can preload some tone zones, to prevent them from getting
# overwritten by other users (if you allow non-root users to open /dev/dahdi/*
# interfaces anyway). Also this means they won't have to be loaded at runtime.
# The format is "loadzone=<zone>" where the zone is a two letter country code.
#
# You may also specify a default zone with "defaultzone=<zone>" where zone
# is a two letter country code.
#
# An up-to-date list of the zones can be found in the file zonedata.c
#
loadzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
#loadzone=hu
#loadzone=lt
#loadzone=pl
defaultzone=us
#
loadzone=uk     # to load the tone set
defaultzone=uk  # to default DAHDI to using that set
cidstart=polarity   ; the delivery of caller ID will be 
                    ; signaled by a polarity reversal
cidsignalling=v23   ; the delivery of the called ID information 
                    ; will be in V23 format
sendcalleridafter=2
language=en_UK
; gives the choice of (1) French, (2) Spanish, or (3) German
exten => s,1,Background(choose-language)
   same => n,WaitExten(5)

exten => 1,1,Set(CHANNEL(language)=fr)

exten => 2,1,Set(CHANNEL(language)=es)

exten => 3,1,Set(CHANNEL(language)=de)

; the next priority for extensions 1, 2, or 3 would be handled here
exten => _[123],n,Goto(menu,s,1)
[zonemessages]
; Users may be located in different timezones, or may have different
; message announcements for their introductory message when they enter
; the voicemail system. Set the message and the timezone each user
; hears here. Set the user into one of these zones with the tz=attribute
; in the options field of the mailbox. Of course, language substitution
; still applies here so you may have several directory trees that have
; alternate language choices.
;
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution is done on the values below.
;
; Supported values:
; 'filename' filename of a soundfile (single ticks around the filename
; required)
; ${VAR} variable substitution
; A or a Day of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...)
; d or e numeric day of month (first, second, ... thirty-first)
; Y Year
; I or l Hour, 12 hour clock
; H Hour, 24 hour clock (single digit hours preceded by "oh")
; k Hour, 24 hour clock (single digit hours NOT preceded by "oh")
; M Minute, with 00 pronounced as "o'clock"
; N Minute, with 00 pronounced as "hundred" (US military time)
; P or p AM or PM
; Q "today", "yesterday" or ABdY
; (*note: not standard strftime value)
; q " (for today), "yesterday", weekday, or ABdY
; (*note: not standard strftime value)
; R 24 hour time, including minute
;
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM

UK24=Europe/London|'vm-received' q 'digits/at' H N 'hours'
UK12=Europe/London|'vm-received' Q 'digits/at' IMp    
[default]
4001 => 1234,Russell Bryant,rb@shifteight.org,,|tz=central
4002 => 4444,David Duffett,dd@shifteight.org,,|tz=UK24
4003 => 4450,Mary Poppins,mp@shifteight.org,,|tz=UK12|attach=yes
;
; indications.conf
;
; Configuration file for location specific tone indications
;
; NOTE:
; When adding countries to this file, please keep them in alphabetical
; order according to the 2-character country codes!
;
; The [general] category is for certain global variables.
; All other categories are interpreted as location specific indications
;
[general]
country=uk   ; default is US, so we have changed it to UK
$[expression]
$[${COUNT} + 1]
$[${COUNT} / 2]
exten => 321,1,Set(COUNT=3)
    same => n,Set(NEWCOUNT=$[${COUNT} + 1])
    same => n,SayNumber(${NEWCOUNT})
exten => 321,n,Set(NEWCOUNT=$[3 + 1])
exten => 321,n,Set(NEWCOUNT=4)
exten => 123,1,Set(TEST=$[2+1])
exten => 234,1,Set(TEST=$[2 + 1])
exten => 234,1,Set(NEWTEST=blah${TEST})
FUNCTION_NAME(argument)
${FUNCTION_NAME(argument)}
${FUNCTION_NAME(${FUNCTION_NAME(argument)})}
 ^             ^ ^             ^        ^^^^
 1             2 3             4        4321
exten => 123,1,Set(TEST=example)
    same => n,SayNumber(${LEN(${TEST})})
exten => s,1,Set(TIMEOUT(digit)=30)
GotoIf(expression?destination1:destination2)
exten => 345,1,Set(TEST=1)
   same => n,GotoIf($[${TEST} = 1]?weasels:iguanas)
   same => n(weasels),Playback(weasels-eaten-phonesys)
   same => n,Hangup()
   same => n(iguanas),Playback(office-iguanas)
   same => n,Hangup()
exten => 345,1,Set(TEST=1)
   same => n,GotoIf($[${TEST} = 1]?:iguanas) ; we don't have the weasels label anymore,
                                             ; but this will still work
   same => n,Playback(weasels-eaten-phonesys)
   same => n,Hangup()
   same => n(iguanas),Playback(office-iguanas)
   same => n,Hangup()
exten => 345,1,Set(TEST=1)
   same => n,GotoIf($[${TEST} = 1]?weasels,1:iguanas,1) ; now we're going to
                                                        ; extension,priority

exten => weasels,1,Playback(weasels-eaten-phonesys)     ; this is NOT a label.
                                                        ; It is a different extension
   same => n,Hangup()

exten => iguanas,1,Playback(office-iguanas)
   same => n,Hangup()
exten => 123,1,Set(COUNT=10)
   same => n(start),GotoIf($[${COUNT} > 0]?:goodbye)
   same => n,SayNumber(${COUNT})
   same => n,Set(COUNT=$[${COUNT} - 1])
   same => n,Goto(start)
   same => n(goodbye),Hangup()
exten => 123,1,GotoIf($[${CALLERID(num)} = 8885551212]?reject:allow)
   same => n(allow),Dial(DAHDI/4)
   same => n,Hangup()
   same => n(reject),Playback(abandon-all-hope)
   same => n,Hangup()
GotoIfTime(times,days_of_week,days_of_month,months?label)
exten => s,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,s,1)
; If it's any hour of the day, on any day of the week,
; during the fourth day of the month, in the month of July,
; we're closed
exten => s,1,GotoIfTime(*,*,4,jul?closed,s,1)

; During business hours, send calls to the open context
   same => n,GotoIfTime(09:00-17:59,mon-fri,*,*?open,s,1)
   same => n,GotoIfTime(09:00-11:59,sat,*,*?open,s,1)

; Otherwise, we're closed
   same => n,Goto(closed,s,1)
exten => 101,1,Dial(${JOHN},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(101@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(101@default,b)
   same => n,Hangup()
[macro-voicemail]
[macro-voicemail]
exten => s,1,Dial(${JOHN},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(101@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(101@default,b)
   same => n,Hangup()
exten => 101,1,Macro(voicemail)
[macro-voicemail]
exten => s,1,Dial(${JOHN},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(${MACRO_EXTEN}@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(${MACRO_EXTEN}@default,b)
   same => n,Hangup()
[macro-voicemail]
exten => s,1,Dial(${ARG1},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(${MACRO_EXTEN}@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(${MACRO_EXTEN}@default,b)
   same => n,Hangup()
exten => 101,1,Macro(voicemail,${JOHN})
exten => 102,1,Macro(voicemail,${JANE})
exten => 103,1,Macro(voicemail,${JACK})
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
   same => n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,VoiceMail(${MACRO_EXTEN},u)
   same => n,Goto(incoming,s,1)

exten => s-BUSY,1,VoiceMail(${MACRO_EXTEN},b)
   same => n,Goto(incoming,s,1)

exten => _s-.,1,Goto(s-NOANSWER,1)
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
   same => n,VoiceMail(${MACRO_EXTEN},${IF($[${DIALSTATUS} = BUSY]?b:u)})
[subVoicemail]
[macro-voicemail]
exten => s,1,Dial(${JOHN},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(101@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(101@default,b)
   same => n,Hangup()
[subVoicemail]
exten => start,1,Dial(${JOHN},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(101@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(101@default,b)
   same => n,Hangup()
exten => 101,1,GoSub(subVoicemail,start,1())
[subVoicemail]
exten => start,1,Dial(${ARG1},10)
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   same => n(unavail),VoiceMail(${ARG2}@default,u)
   same => n,Hangup()
   same => n(busy),VoiceMail(${ARG2}@default,b)
   same => n,Hangup()
[LocalSets]
exten => 101,1,GoSub(subVoicemail,start,1(${JOHN},${EXTEN}))
exten => 102,1,GoSub(subVoicemail,start,1(${JANE},${EXTEN}))
exten => 103,1,GoSub(subVoicemail,start,1(${JACK},${EXTEN}))
[subVoicemail]
exten => start,1,Dial(${ARG1},10)
   same => n,VoiceMail(${ARG2}@default,${IF($[${DIALSTATUS} = BUSY]?b:u)})
   same => n,Hangup()
[subDialer]
exten => start,1,Dial(${ARG1},${ARG2})
   same => n,Return()

[subVoicemail]
exten => start,1,VoiceMail(${ARG1}@${ARG2},${ARG3})
   same => n,Hangup()
exten => 101,1,GoSub(subDialer,start,1(${JOHN},30))
   same => n,GoSub(subVoicemail,start,1(${EXTEN},default,u))
[subDialer]
exten => start,1,Dial(${ARG1},${ARG2})
   same => n,Return(${DIALSTATUS})

[subVoicemail]
exten => start,1,VoiceMail(${ARG1}@${ARG2},${ARG3})
   same => n,Hangup()
exten => 101,1,GoSub(subDialer,start,1(${JOHN},30))
   same => n,Set(VoicemailMessage=${IF($[${GOSUB_RETVAL} = BUSY]?b:u)})
   same => n,GoSub(subVoicemail,start,1(${EXTEN},default,${VoicemailMessage}))
[LocalSets]
exten => 107,1,Verbose(2,Dialing multiple locations simultaneously)
   same => n,Dial(SIP/0000FFFF0001&DAHDI/g0/14165551212&SIP/MyITSP/12565551212,30)
   same => n,Hangup()
[LocalSets]
exten => 107,1,Verbose(2,Dialing multiple locations with time delay)

; *** This all needs to be on a single line
   same => n,Dial(Local/channel_1@TimeDelay&Local/channel_2@TimeDelay
&Local/channel_3@TimeDelay,40)
   same => n,Hangup()
[LocalSets]
exten => 107,1,Verbose(2,Dialing multiple locations with time delay)

; *** This all needs to be on a single line
   same => n,Dial(Local/channel_1@TimeDelay&Local/channel_2@TimeDelay
&Local/channel_3@TimeDelay,40)
   same => n,Hangup()

[TimeDelay]
exten => channel_1,1,Verbose(2,Dialing the first channel)
   same => n,Dial(SIP/0000FFFF0001,20)
   same => n,Hangup()

exten => channel_2,1,Verbose(2,Dialing the second channel with a delay)
   same => n,Wait(10)
   same => n,Dial(DAHDI/g0/14165551212)

exten => channel_3,1,Verbose(2,Dialing the third channel with a delay)
   same => n,Wait(15)
   same => n,Dial(SIP/MyITSP/12565551212,15)
   same => n,Hangup()
exten => 456,1,Set(DB(test/count)=1)
exten => 456,1,Set(DB(test/count)=1)
   same => n,Set(COUNT=${DB(test/count)})
   same => n,SayNumber(${COUNT})
; deletes the key and returns its value in one step
exten => 457,1,Verbose(0, The value was ${DB_DELETE(test/count)})
exten => 457,1,DBdeltree(test)
exten => 678,1,Set(COUNT=${DB(test/count)})
   same => n,GotoIf($[${ISNULL(${COUNT})}]?:continue)
   same => n,Set(DB(test/count)=1)
   same => n,Goto(1)
   same => n(continue),NoOp()
exten => 678,1,Set(COUNT=${DB(test/count)})
   same => n,GotoIf($[${ISNULL(${COUNT})}]?:continue)
   same => n,Set(DB(test/count)=1)
   same => n,Goto(1)
   same => n(continue),NoOp()
   same => n,SayNumber(${COUNT})
   same => n,Set(COUNT=$[${COUNT} + 1])
exten => 678,1,Set(COUNT=${DB(test/count)})
   same => n,GotoIf($[${ISNULL(${COUNT})}]?:continue)
   same => n,Set(DB(test/count)=1)
   same => n,Goto(1)
   same => n(continue),NoOp()
   same => n,SayNumber(${COUNT})
   same => n,Set(COUNT=$[${COUNT} + 1])
   same => n,Set(DB(test/count)=${COUNT})
exten => 678,1,Set(COUNT=${DB(test/count)})
   same => n,GotoIf($[${ISNULL(${COUNT})}]?:continue)
   same => n,Set(DB(test/count)=1)
   same => n,Goto(1)
   same => n(continue),NoOp()
   same => n,SayNumber(${COUNT})
   same => n,Set(COUNT=$[${COUNT} + 1]
   same => n,Set(DB(test/count)=${COUNT})
   same => n,Goto(1)
exten => 124,1,GotoIf($[${BLACKLIST()]?blocked,1)
   same => n,Dial(${JOHN})

exten => blocked,1,Playback(privacy-you-are-blacklisted)
   same => n,Playback(vm-goodbye)
   same => n,Hangup()
[incomimg]
exten => s,1,Zapateller(nocallerid)
   same => n,Playback(enter-ext-of-person)
[incoming]
include => parkedcalls

exten => 103,1,Dial(SIP/Bob,,tT)
exten => 104,1,Dial(SIP/Charlie,,tT)
[rooms]
conf => 600
exten => 600,1,MeetMe(600,i,54321)
exten => 601,1,Playback(conf-thereare)
   same => n,MeetMeCount(600)
   same => n,Playback(conf-peopleinconf)
; limit the conference room to 10 participants
exten => 600,1,MeetMeCount(600,CONFCOUNT)
   same => n,GotoIf($[${CONFCOUNT} <= 10]?meetme:conf_full,1)
   same => n(meetme),MeetMe(600,i,54321)

exten => conf_full,1,Playback(conf-full)
<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>]
,<Application>([<AppArguments>])[,MOH_Class]
agi_test => *6,self/callee,AGI(agi-test.agi),default
exten => 101,n,Set(__DYNAMIC_FEATURES=agi_test)
exten => 101,n,Dial(SIP/0000FFFF0002)

*CLI> features reload
exten => example,1,Set(_MyVariable=thisValue)
exten => example,1,Set(__MyVariable=thisValue)
exten => example,1,Verbose(1,Value of MyVariable is: ${MyVariable})
[shifteight]
unpauseMonitor => *1    ; custom key mapping
pauseMonitor => *2      ; custom key mapping
agi_test =>             ; no custom key mapping

Set(__DYNAMIC_FEATURES=shifteight) ; use the double underscore if you want to ensure
                                   ; both call legs have the variable assigned.

parkext => 700      ; What extension to dial to park (all parking lots)
parkpos => 701-720  ; What extensions to park calls on (defafult parking lot)
exten => *724,1,Page(${ChannelsToPage},i,120)
exten => *724,1,Verbose(2,Paging to external amplifier) ; note the '*' in the 
                                                        ; extension is part of 
                                                        ; what you actually dial
   same => n,Set(PageDevice=SIP/PagingATA)
   same => n,Page(${PageDevice},i,120)
   same => n,Dial(DAHDI/25)
Page(SIP/SET1&SIP/SET2&SIP/SET3&SIP/SET4&SIP/SET5&SIP/SET6&SIP/SET7&...
exten => *724,1,Verbose(2,Paging to Aastra sets)
   same => n,SIPAddHeader(Alert-Info: info=alert-autoanswer)
   same => n,Set(PageDevice=SIP/00085D000000)
   same => n,Page(${PageDevice},i)
exten => *724,1,Verbose(2,Paging to Polycom sets)
   same => n,SIPAddHeader(Alert-Info: Ring Answer)
   same => n,Set(PageDevice=SIP/0004F2000000)
   same => n,Page(${PageDevice},i)
exten => *724,1,Verbose(2,Paging to Snom sets)
   same => n,Set(VXML_URL=intercom=true)

; replace 'domain.com' with the domain of your system
   same => n,SIPAddHeader(Call-Info: sip:domain.com\;answer-after=0)
   same => n,Set(PageDevice=SIP/000413000000)
   same => n,Page(${PageDevice},i)
exten => *724,1,Verbose(2,Paging to Cisco SPA sets -- but not Cisco 79XX sets)
   same => n,SIPAddHeader(Call-Info:\;answer-after=0)        ; Cisco SPA phones
   same => n,Set(PageDevice=SIP/0004F2000000)
   same => n,Page(${PageDevice},i)
MulticastRTP/<type>/<ip address:port>[/<linksys address:port>]
exten => *723,1,Page(MulticastRTP/basic/239.0.0.1:1234)
exten => *724,1,Page(MulticastRTP/linksys/239.0.0.1:1234/239.0.0.1:6061)
[global]

MULTICAST=MulticastRTP/linksys/239.0.0.1:1234
;MULTICAST=MulticastRTP/linksys/239.0.0.1:1234/239.0.0.1:6061 ; if you have SPA 
                                                              ; (Linksys/Cisco)
                                                              ; phones

BOGEN=SIP/ATAforPaging  ; This assumes an ATA in your sip.conf file named 
                        ; [ATAforPaging]
;BOGEN=DAHDI/25         ; We could do this too, assuming we have an analog 
                        ; FXS card at DAHDI channel 25
PAGELIST=${MULTICAST}&${BOGEN} ; All of these variable names are arbitrary. 
                               ; Asterisk doesn't care what you call these strings

[page_context] ; You don't need a page context, so long as the extension you 
               ; assign to paging is dialable by your sets

exten => *724,1,Page(${PAGELIST},i,120)
        _sip._udp.shifteight.org. 86400 IN SRV 0 0 5060 pbx.shifteight.org.

# dig SRV _sip._udp.shifteight.org
        ;; ANSWER SECTION:
_sip._udp.shifteight.org. 14210 IN SRV 0 0 5060 pbx.shifteight.org.

[general]
context=unauthenticated         ; default context for incoming calls
allowguest=yes                  ; enable unauthenticated calls

$ sudo asterisk -rx "sip reload"
*CLI> sip reload
[unauthenticated]
exten => leif,1,Goto(PublicExtensions,100,1)

exten => jim,1,Goto(PublicExtensions,101,1)

exten => tilghman,1,Goto(PublicExtensions,102,1)

exten => russell,1,Goto(PublicExtensions,103,1)

[unauthenticated]
exten => _[A-Za-z0-9].,1,Verbose(2,UNAUTHENTICATED REQUEST TO ${EXTEN} FROM 
${CALLERID(all)})
   same => n,Set(FilteredExtension=${FILTER(A-Za-z0-9,${EXTEN})})
   same => n,Set(CheckPublicExtensionResult=${DIALPLAN_EXISTS(PublicExtensions,
${FilteredExtension},1)})
   same => n,GotoIf($["${CheckPublicExtensionResult}" = "0"]?CheckEmailLookup)
   same => n,Goto(PublicExtensions,${FilteredExtension},1)

; This is our handler for when someone dials a SIP URI with a name
   same => n(CheckEmailLookup),GoSub(subEmailToExtensionLookup,start,1
(${TOLOWER(${FilteredExtension})}))
   same => n,GotoIf($["${GOSUB_RETVAL}" = "NoResult"]?i,1:PublicExtensions,
${GOSUB_RETVAL},1)
   same => n,Goto(i,1)

; This handles invalid numbers/names
exten => i,1,Verbose(2,Incoming call from ${CALLERID(all)} to context ${CONTEXT} 
found no result)
   same => n,Playback(silence/1&invalid)
   same => n,Hangup()

; These are explicit extension matches (useful on small systems)
exten => leif,1,Goto(PublicExtensions,100,1)

exten => jim,1,Goto(PublicExtensions,101,1)

exten => tilghman,1,Goto(PublicExtensions,102,1)

exten => russell,1,Goto(PublicExtensions,103,1)

; where 'name' is the username as found in the email address
GoSub(subEmailToExtensionLookup,start,1(name))
[subEmailToExtensionLookup]
exten => start,1,Verbose(2,Checking for user in voicemail.conf)
   same => n,Set(LOCAL(FilteredExtension)=${FILTER(a-z0-9,${ARG1})})
   same => n,Set(LOCAL(Result)=${SHELL(grep "${LOCAL(FilteredExtension)}@" 
/etc/asterisk/voicemail.conf)})
   same => n,GotoIf($[${ISNULL(${LOCAL(Result)})}]?no_Result,1)
   same => n,Set(LOCAL(ExtensionToDial)=${CUT(${LOCAL(Result)},=,1)})
   same => n,Set(LOCAL(ExtensionToDial)=${FILTER(0-9,${LOCAL(ExtensionToDial)})})
   same => n,Return(${LOCAL(ExtensionToDial)})

exten => no_Result,1,Verbose(2,No user ${ARG1} found in voicemail.conf)
   same => n,Return(NoResult)

Set(LOCAL(FilteredExtension)=${FILTER(a-z0-9,${ARG1})})
Set(LOCAL(Result)=${SHELL(grep "${LOCAL(FilteredExtension)}@" /etc/asterisk/voicemail.conf)})
GotoIf($[${ISNULL(${LOCAL(Result)})}]?no_result,1)
exten => no_result,1,Verbose(2,No user ${ARG1} found in voicemail.conf)
   same => n,Return(NoResult)
Set(LOCAL(ExtensionToDial)=${CUT(${LOCAL(Result)},=,1)})
Set(LOCAL(ExtensionToDial)=${FILTER(0-9,${LOCAL(ExtensionToDial)})})
Return(${LOCAL(ExtensionToDial)})
[subLookupNameInNameMappingTable]
exten => start,1,Verbose(2,Looking up ${ARG1})

; where 'name' is the username as found in the email address 
   same => n,Set(ARRAY(CalleeExtension,CalleeContext)=${GET_NAME_LOOKUP(${ARG1})})
   same => n,GotoIf($[${ISNULL(${CalleeExtension})}]?no_result,1)
   same => n,GotoIf($[${ISNULL(${CalleeContext})}]?no_result,1)
   same => n,Return() ; You'll need to handle the new CalleeExtension and 
                      ; CalleeContext variables in the code that called this 
                      ; subroutine

exten => no_result,1,Verbose(2,Name was not found in the database.)
   same => n,Return(NoResult)

[NAME_LOOKUP](DB)
prefix=GET
SELECT Extension,Context FROM NameMapping WHERE Name='${ARG1}'

5-444-6-*-888-2-66(pause)-6-33-4(pause)-4-33-555-33-66-#-7777-44(pause)-444-333-8-33-
444-(pause)-4(pause)-44-8-*-666-777-4
exten => _[0-9a-zA-Z].,1,Verbose()
   same => n,Set(FilteredExtension=${FILTER(0-9a-zA-Z@-_.,${EXTEN})})
   same => n,Dial(SIP/${FilteredExtension})
exten => 5343,1,Dial(SIP/leif.madsen@shifteight.org)
_NXXNXXXXXX
_1NXXNXXXXXX
_011X.
_N11

_0[123789]XXXXXXXXX
_0[123789]XXXXXXXX

_NXXXXXXX
_0XXXXXXXXX

1. Remove all characters with the exception of the digits.
For example, the First Well Known Rule produced the Key
"+442079460148".  This step would simply remove the
leading "+", producing "442079460148".

2. Put dots (".") between each digit. Example:
      4.4.2.0.7.9.4.6.0.1.4.8

3. Reverse the order of the digits. Example:
      8.4.1.0.6.4.9.7.0.2.4.4

4. Append the string ".e164.arpa" to the end.  Example:
      8.4.1.0.6.4.9.7.0.2.4.4.e164.arpa

exten => _X.,1,Set(CurrentExten=${FILTER(0-9,${EXTEN})})
   same => n,Set(LookupResult=${ENUMLOOKUP(${CurrentExten},sip,,,e164.arpa)})
   same => n,GotoIf($[${EXISTS(${LookupResult})}]?HaveLocation,1)
   same => n,Set(LookupResult=${ENUMLOOKUP(${CurrentExten},sip,,,e164.org)})
   same => n,GotoIf($[${ISNULL(${LookupResult})}]?NormalCall,1:HaveLocation,1)

exten => HaveLocation,1,Verbose(2,Handle dialing via SIP URI returned)
   exten => ...

exten => NormalCall,1,Verbose(2,Handle dialing via standard PSTN route)
   exten => ...

0*1273
!^\\+*([^\\*]*)!sip:\\1@sip.yourdomain.com!
!^\\+*([^\\*]*)!sip:\\1@pbx.shifteight.org!
$ dig NAPTR 4.3.2.1.1273.freenum.org
;; ANSWER SECTION: 
4.3.2.1.1273.freenum.org. 86400 IN NAPTR 100 10 "u" "E2U+sip" 
"!^\\+*([^\\*]*)!sip:\\1@shifteight.org!" .
[globals]
ITAD = 1273 ; replace '1273' with your own ITAD number
[OutgoingISN]
exten => _X*X!,1,GoSub(subFreenum,start,1(${EXTEN}))
exten => _XX*X!,1,GoSub(subFreenum,start,1(${EXTEN}))
exten => _XXX*X!,1,GoSub(subFreenum,start,1(${EXTEN}))
exten => _XXXX*X!,1,GoSub(subFreenum,start,1(${EXTEN}))
exten => _XXXXX*X!,1,GoSub(subFreenum,start,1(${EXTEN}))
; you may need to add more lines here to handle XXXXXX*X, XXXXXXX*X, and so forth

[subFreenum]
exten => start,1,Verbose(2,Performing ISN lookup)
   same => n,Set(ISN=${FILTER(0-9*,${ARG1})})
   same => n,Set(Result=${ENUMLOOKUP(${ISN},sip,s,,freenum.org)})
   same => n,GotoIf($[${EXISTS(${Result})}]?call,1:no_result,1)

exten => call,1,Verbose(2,Placing call to ISN --${ISN}-- via ${Result})
   same => n,Dial(SIP/${Result})
   same => n,Return()

exten => no_result,1,Verbose(2,Lookup for ISN: --${ISN}-- returned no result)
   same => n,Playback(silence/1&invalid)
   same => n,Return()

[LocalSets]
include => OutgoingISN  ; include the context that enables ISN dialing
include => CallPlace    ; use subroutine to determine what you can dial

exten => start,n,Set(ISN=${FILTER(0-9*,${ARG1})})
exten => start,n,Set(Result=${ENUMLOOKUP(${ISN},sip,s,,freenum.org)})
exten => start,n,GotoIf($[${EXISTS(${Result})}]?call,1:no_result,1)
$ cd /etc/asterisk/
$ touch queues.conf
[general]
autofill=yes             ; distribute all waiting callers to available members
shared_lastcall=yes      ; respect the wrapup time for members logged into more 
                         ; than one queue

[StandardQueue](!)       ; template to provide common features
musicclass=default       ; play [default] music
strategy=rrmemory        ; use the Round Robin Memory strategy
joinempty=no             ; do not join the queue when no members available
leavewhenempty=yes       ; leave the queue when no members available
ringinuse=no             ; don't ring members when already InUse (prevents 
                         ; multiple calls to an agent)

[sales](StandardQueue)   ; create the sales queue using the parameters in the
                         ; StandardQueue template

[support](StandardQueue) ; create the support queue using the parameters in the
                         ; StandardQueue template
$ asterisk -r
*CLI> module reload app_queue.so
   -- Reloading module 'app_queue.so' (True Call Queueing)
localhost*CLI> queue show
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   No Members
   No Callers

sales        has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   No Members
   No Callers
[Queues]
exten => 7001,1,Verbose(2,${CALLERID(all)} entering the support queue)
same => n,Queue(support)
same => n,Hangup()

exten => 7002,1,Verbose(2,${CALLERID(all)} entering the sales queue)
same => n,Queue(sales)
same => n,Hangup()

[LocalSets]
include => Queues      ; allow phones to call queues
 -- Executing [7001@LocalSets:1] Verbose("SIP/0000FFFF0003-00000001", 
    "2,"Leif Madsen" <100> entering the support queue") in new stack
== "Leif Madsen" <1--> entering the support queue
 -- Executing [7001@LocalSets:2] Queue("SIP/0000FFFF0003-00000001", 
    "support") in new stack
    [2011-02-14 08:59:39] WARNING[13981]: app_queue.c:5738 queue_exec: 
    Unable to join queue 'support'
 -- Executing [7001@LocalSets:3]
    Hangup("SIP/0000FFFF0003-00000001", "") in new stack
 == Spawn extension (LocalSets, 7001, 3) exited non-zero on 
    'SIP/0000FFFF0003-00000001'
*CLI> queue add member <channel> to <queue> [[[penalty <penalty>] as 
<membername>] state_interface <interface>]
*CLI> queue add member SIP/0000FFFF0001 to support
Added interface 'SIP/0000FFFF0001' to queue 'support'
*CLI> queue show support
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: 
      SIP/0000FFFF0001 (dynamic) (Not in use) has taken no calls yet
   No Callers
*CLI> queue remove member SIP/0000FFFF0001 from support 
Removed interface 'SIP/0000FFFF0001' from queue 'support'
*CLI> queue pause member SIP/0000FFFF0001 queue support reason DoingCallbacks
paused interface 'SIP/0000FFFF0001' in queue 'support' for reason 'DoingCallBacks'

*CLI> queue show support
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: 
      SIP/0000FFFF0001 (dynamic) (paused) (Not in use) has taken no calls yet
   No Callers
*CLI> queue unpause member SIP/0000FFFF0001 queue support reason off-break
unpaused interface 'SIP/0000FFFF0001' in queue 'support' for reason 'off-break'

*CLI> queue show support
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: 
      SIP/0000FFFF0001 (dynamic) (Not in use) has taken no calls yet
   No Callers
[QueueMemberFunctions]

exten => *54,1,Verbose(2,Logging In Queue Member)
   same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)})
   same => n,AddQueueMember(support,${MemberChannel})

; ${AQMSTATUS}
;   ADDED
;   MEMBERALREADY
;   NOSUCHQUEUE

exten => *56,1,Verbose(2,Logging Out Queue Member)
   same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)})
   same => n,RemoveQueueMember(support,${MemberChannel})

; ${RQMSTATUS}:
;    REMOVED
;    NOTINQUEUE
;    NOSUCHQUEUE

exten => *72,1,Verbose(2,Pause Queue Member)
   same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)})
   same => n,PauseQueueMember(support,${MemberChannel})

; ${PQMSTATUS}:
;     PAUSED
;     NOTFOUND

exten => *87,1,Verbose(2,Unpause Queue Member)
   same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)})
   same => n,UnpauseQueueMember(support,${MemberChannel})

; ${UPQMSTATUS}:
;     UNPAUSED
;     NOTFOUND
*CLI> database put queue_agent 0000FFFF0001/available_queues support^sales
pbx*CLI> database show queue_agent
/queue_agent/0000FFFF0001/available_queues        : support^sales
[subSetupAvailableQueues]
;
; This subroutine is used by the various login/logout/pausing/unpausing routines
; in the [ACD] context. The purpose of the subroutine is centralize the retrieval 
; of information easier.
;
exten => start,1,Verbose(2,Checking for available queues)

; Get the current channel's peer name (0000FFFF0001)
   same => n,Set(MemberChannel=${CHANNEL(peername)})

; Get the current channel's technology type (SIP, IAX, etc)
   same => n,Set(MemberChanType=${CHANNEL(channeltype)})    

; Get the list of queues available for this agent
   same => n,Set(AvailableQueues=${DB(queue_agent/${MemberChannel}/
   available_queues)})
; *** This should all be on a single line

; if there are no queues assigned to this agent we'll handle it in the 
; no_queues_available extension
   same => n,GotoIf($[${ISNULL(${AvailableQueues})}]?no_queues_available,1) 

   same => n,Return()

exten => no_queues_available,1,Verbose(2,No queues available for agent 
   ${MemberChannel})
; *** This should all be on a single line

; playback a message stating the channel has not yet been assigned
   same => n,Playback(silence/1&channel&not-yet-assigned)  
   same => n,Hangup()

[ACD]
;
; Used for logging agents into all configured queues per the AstDB
;
;
; Logging into multiple queues via the AstDB system
exten => *54,1,Verbose(2,Logging into multiple queues per the database values)

; get the available queues for this channel
   same => n,GoSub(subSetupAvailableQueues,start,1())  
   same => n,Set(QueueCounter=1)  ; setup a counter variable

; using CUT(), get the first listed queue returned from the AstDB
   same => n,Set(WorkingQueue=${CUT(AvailableQueues,^,${QueueCounter})})

; While the WorkingQueue channel variable contains a value, loop
   same => n,While($[${EXISTS(${WorkingQueue})}])

; AddQueueMember(queuename[,interface[,penalty[,options[,membername
;  [,stateinterface]]]]])
; Add the channel to a queue, setting the interface for calling 
; and the interface for monitoring of device state
;
; *** This should all be on a single line
   same => n,AddQueueMember(${WorkingQueue},${MemberChanType}/
${MemberChannel},,,${MemberChanType}/${MemberChannel})

   same => n,Set(QueueCounter=$[${QueueCounter} + 1])    ; increase our counter

; get the next available queue; if it is null our loop will end
   same => n,Set(WorkingQueue=${CUT(AvailableQueues,^,${QueueCounter})})

   same => n,EndWhile()

; let the agent know they were logged in okay
   same => n,Playback(silence/1&agent-loginok)
   same => n,Hangup()

exten => no_queues_available,1,Verbose(2,No queues available for ${MemberChannel})
   same => n,Playback(silence/1&channel&not-yet-assigned)
   same => n,Hangup()

; -------------------------

; Used for logging agents out of all configured queues per the AstDB
exten => *56,1,Verbose(2,Logging out of multiple queues)

; Because we reused some code, we've placed the duplicate code into a subroutine
   same => n,GoSub(subSetupAvailableQueues,start,1())   
   same => n,Set(QueueCounter=1)
   same => n,Set(WorkingQueue=${CUT(AvailableQueues,^,${QueueCounter})})
   same => n,While($[${EXISTS(${WorkingQueue})}])
   same => n,RemoveQueueMember(${WorkingQueue},${MemberChanType}/${MemberChannel})
   same => n,Set(QueueCounter=$[${QueueCounter} + 1])
   same => n,Set(WorkingQueue=${CUT(AvailableQueues,^,${QueueCounter})})
   same => n,EndWhile()
   same => n,Playback(silence/1&agent-loggedoff)
   same => n,Hangup()

; -------------------------

; Used for pausing agents in all available queues
exten => *72,1,Verbose(2,Pausing member in all queues)
   same => n,GoSub(subSetupAvailableQueues,start,1())

   ; if we don't define a queue, the member is paused in all queues
   same => n,PauseQueueMember(,${MemberChanType}/${MemberChannel})
   same => n,GotoIf($[${PQMSTATUS} = PAUSED]?agent_paused,1:agent_not_found,1)

exten => agent_paused,1,Verbose(2,Agent paused successfully)
   same => n,Playback(silence/1&unavailable)
   same => n,Hangup()

; -------------------------

; Used for unpausing agents in all available queues
exten => *87,1,Verbose(2,UnPausing member in all queues)
   same => n,GoSub(subSetupAvailableQueues,start,1())

   ; if we don't define a queue, then the member is unpaused from all queues
   same => n,UnPauseQueueMember(,${MemberChanType}/${MemberChannel})
   same => n,GotoIf($[${UPQMSTATUS} = UNPAUSED]?agent_unpaused,1:agent_not_found,1)

exten => agent_unpaused,1,Verbose(2,Agent paused successfully)
   same => n,Playback(silence/1&available)
   same => n,Hangup()

; -------------------------

; Used by both pausing and unpausing dialplan functionality
exten => agent_not_found,1,Verbose(2,Agent was not found)
   same => n,Playback(silence/1&cannot-complete-as-dialed)
*CLI> queue show support
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: 
      SIP/0000FFFF0001 (dynamic) (Not in use) has taken no calls yet
   No Callers
    -- Executing [555@LocalSets:1] MusicOnHold("SIP/0000FFFF0001-00000000", 
       "") in new stack
    -- Started music on hold, class 'default', on SIP/0000FFFF0001-00000000

*CLI> queue show support
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: 
      SIP/0000FFFF0001 (dynamic) (Not in use) has taken no calls yet
   No Callers
[general]
context=unauthenticated         ; default context for incoming calls
allowguest=no                   ; disable unauthenticated calls
srvlookup=yes                   ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0             ; listen for UDP request on all interfaces
tcpenable=no                    ; disable TCP support
callcounter=yes                 ; enable device states for SIP devices
*CLI> sip reload
 Reloading SIP 
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Using SIP RTP CoS mark 5
    -- Executing [555@LocalSets:1] MusicOnHold("SIP/0000FFFF0001-00000001", 
       "") in new stack
    -- Started music on hold, class 'default', on SIP/0000FFFF0001-00000001

*CLI> queue show support
support      has 0 calls (max unlimited) in 'rrmemory' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: 
      SIP/0000FFFF0001 (dynamic) (In use) has taken no calls yet
   No Callers
agent => agent_id,agent_password,name
agent => 1000,1234,Danielle Roberts
[support_template](!)
musicclass=default
strategy=rrmemory
joinempty=no
leavewhenempty=yes
ringinuse=no

[support](support_template)
weight=0

[support-priority](support_template)
weight=10
[LocalSets]
include => Queue  ; allow direct transfer of calls to queues

[Queues]
exten => 7000,1,Verbose(2,Entering the support queue)
   same => n,Queue(support)             ; standard support queue available
                                        ; at extension 7000
   same => n,VoiceMail(7000@queues,u)   ; if there are no members in the queue, 
                                        ; we exit and send the caller to voicemail
   same => n,Hangup()

exten => 8000,1,Verbose(2,Entering the priority support queue)
   same => n,Queue(support-priority)    ; priority queue available at 
                                        ; extension 8000
   same => n,VoiceMail(7000@queues,u)   ; if there are no members in the queue, 
                                        ; we exit and send the caller to voicemail
   same => n,Hangup()
[support](StandardQueue)
member => SIP/0000FFFF0001,0,James Shaw         ; preferred
member => SIP/0000FFFF0002,10,Kay Madsen        ; second preferred
member => SIP/0000FFFF0003,20,Danielle Roberts  ; least preferred

[sales](StandardQueue)
member => SIP/0000FFFF0002,0,Kay Madsen
member => SIP/0000FFFF0003,10,Danielle Roberts
member => SIP/0000FFFF0001,20,James Shaw

[billing](StandardQueue)
member => SIP/0000FFFF0003,0,Danielle Roberts
member => SIP/0000FFFF0001,10,James Shaw
member => SIP/0000FFFF0002,20,Kay Madsen
exten => *54,1,Verbose(2,Logging In Queue Member)
   same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)})

; *CLI> database put queue support/0000FFFF0001/penalty 0
   same => n,Set(QueuePenalty=${DB(queue/support/${CHANNEL(peername)}/penalty)})

; *CLI> database put queue support/0000FFFF0001/membername "James Shaw"
   same => n,Set(MemberName=${DB(queue/support/${CHANNEL(peername)}/membername)})

; AddQueueMember(queuename[,interface[,penalty[,options[,membername
; [,stateinterface]]]]])
   same => n,AddQueueMember(support,${MemberChannel},${QueuePenalty},,${MemberName})

[Queues]
exten => 7000,1,Verbose(2,Entering the support queue)
   same => n,Set(QUEUE_MIN_PENALTY=2)   ; set minimum queue member penalty to be used
   same => n,Set(QUEUE_MAX_PENALTY=4)   ; set maximum queue member penalty we'll use
   same => n,Queue(support)             ; entering the queue with minimum and maximum 
                                        ; member penalties to be used
[more_members]
penaltychange => 60,5,1
*CLI> module reload app_queue.so
    -- Reloading module 'app_queue.so' (True Call Queueing)
  == Parsing '/etc/asterisk/queuerules.conf':   == Found
*CLI> queue show rules
Rule: more_members
   After 60 seconds, adjust QUEUE_MAX_PENALTY to 5 and adjust QUEUE_MIN_PENALTY to 1
[Queues]
exten => 7000,1,Verbose(2,Entering the support queue)
   same => n,Set(QUEUE_MIN_PENALTY=2)   ; set minimum queue member penalty
   same => n,Set(QUEUE_MAX_PENALTY=4)   ; set maximum queue member penalty

; Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro
; [,gosub[,rule[,position]]]]]]]]])
   same => n,Queue(support,,,,,,,,more_members)  ; entering queue with minimum and 
                                                 ; maximum member penalties
[more_members]
penaltychange => 30,+1
penaltychange => 45,,-1
penaltychange => 60,+1
penaltychange => 120,+2
*CLI> queue show rules
Rule: more_members
  After 30 seconds, adjust QUEUE_MAX_PENALTY by 1 and adjust QUEUE_MIN_PENALTY by 0
  After 45 seconds, adjust QUEUE_MAX_PENALTY by 0 and adjust QUEUE_MIN_PENALTY by -1
  After 60 seconds, adjust QUEUE_MAX_PENALTY by 1 and adjust QUEUE_MIN_PENALTY by 0
  After 120 seconds, adjust QUEUE_MAX_PENALTY by 2 and adjust QUEUE_MIN_PENALTY by 0
[general]
autofill=yes             ; distribute all waiting callers to available members
shared_lastcall=yes      ; respect the wrapup time for members logged into more 
                         ; than one queue

[StandardQueue](!)       ; template to provide common features
musicclass=default       ; play [default] music
strategy=rrmemory        ; use the Round Robin Memory strategy
joinempty=yes            ; do not join the queue when no members available
leavewhenempty=no        ; leave the queue when no members available
ringinuse=no             ; don't ring members when already InUse (prevents 
                         ; multiple calls to an agent)

[sales](StandardQueue)   ; create the sales queue using the parameters in the
                         ; StandardQueue template

[support](StandardQueue) ; create the support queue using the parameters in the
                         ; StandardQueue template

[StandardQueue](!)       ; template to provide common features
musicclass=default       ; play [default] music
strategy=rrmemory        ; use the Round Robin Memory strategy
joinempty=yes            ; do not join the queue when no members available
leavewhenempty=no        ; leave the queue when no members available
ringinuse=no             ; don't ring members when already InUse (prevents 
                         ; multiple calls to an agent)

; -------- Announcement Control --------
announce-frequency=30           ; announces caller's hold time and position every 30
                                ; seconds
min-announce-frequency=30       ; minimum amount of time that must pass before the
                                ; caller's position is announced 
periodic-announce-frequency=45  ; defines how often to play a periodic announcement to
                                ; caller
random-periodic-announce=no     ; defines whether to play periodic announcements in 
                                ; a random order, or serially
relative-periodic-announce=yes  ; defines whether the timer starts at the end of
                                ; file playback (yes) or the beginning (no)
announce-holdtime=once          ; defines whether the estimated hold time should be 
                                ; played along with the periodic announcement
announce-position=limit         ; defines if we should announce the caller's position
                                ; in the queue
announce-position-limit=10      ; defines the limit value where we announce the
                                ; caller's position (when announce-position is set to 
                                ; limit or more)
announce-round-seconds=30       ; rounds the hold time announcement to the nearest 
                                ; 30-second value
[Queues]
exten => 7000,1,Verbose(2,Joining the support queue for a maximum of 2 minutes)
   same => n,Queue(support,120)
   same => n,VoiceMail(support@queues,u)
   same => n,Hangup()
joinempty=paused,inuse,invalid
leavewhenempty=inuse,ringing
; queues.conf
[support](StandardQueue)
member => Local/SIP-0000FFFF0001@MemberConnector  ; pass the technology to dial over
                                                  ; and the device identifier,
                                                  ; separated by a hyphen. We'll
                                                  ; break it apart inside the 
                                                  ; MemberConnector context.
[MemberConnector]
exten => _[A-Za-z0-9].,1,Verbose(2,Connecting ${CALLERID(all)} to Agent at ${EXTEN})

   ; filter out any bad characters, allowing alphanumeric characters and the hyphen
   same => n,Set(QueueMember=${FILTER(A-Za-z0-9\-,${EXTEN})

   ; assign the first field of QueueMember to Technology using the hyphen separator
   same => n,Set(Technology=${CUT(QueueMember,-,1)})

   ; assign the second field of QueueMember to Device using the hyphen separator
   same => n,Set(Device=${CUT(QueueMember,-,2)})

   ; dial the agent
   same => n,Dial(${Technology}/${Device})
   same => n,Hangup()
; queues.conf
[support](StandardQueue)
member => Local/SIP-0000FFFF0001@MemberConnector,,,SIP/0000FFFF0001
[QueueMemberLogin]
exten => 500,1,Verbose(2,Logging in device ${CHANNEL(peername)} into the support queue)

   ; Save the device's technology to the MemberTech channel variable
   same => n,Set(MemberTech=${CHANNEL(channeltype)})

   ; Save the device's identifier to the MemberIdent channel variable
   same => n,Set(MemberIdent=${CHANNEL(peername)})

   ; Build up the interface name and assign it to the Interface channel variable
   same => n,Set(Interface=${MemberTech}/${MemberIdent})

   ; Add the member to the support queue using a Local channel. We're using the same
   ; format as before, separating the technology and the device indentifier with
   ; a hyphen and passing that information to the MemberConnector context. We then
   ; use the IF() function to determine if the member's technology is SIP and, if so,
   ; to pass back the contents of the Interface channel variable as the value to the
   ; state interface field of the AddQueueMember() application.
   ;
   ; *** This line should not have any line breaks
   same => n,AddQueueMember(support,Local/${MemberTech}-${MemberIdent}
@MemberConnector,,,${IF($[${MemberTech} = SIP]?${Interface})})
   same => n,Playback(silence/1)

   ; Play back either the agent-loginok or agent-incorrect file, depending on what
   ; the AQMSTATUS variable is set to.
   same => n,Playback(${IF($[${AQMSTATUS} = ADDED]?agent-loginok:agent-incorrect)})
   same => n,Hangup()
[MemberConnector]
exten => _[A-Za-z0-9].,1,Verbose(2,Connecting ${CALLERID(all)} to Agent at ${EXTEN})

   ; filter out any bad characters, allowing alphanumeric characters and the hyphen
   same => n,Set(QueueMember=${FILTER(A-Za-z0-9\-,${EXTEN})

   ; assign the first field of QueueMember to Technology using the hyphen separator
   same => n,Set(Technology=${CUT(QueueMember,-,1)})

   ; assign the second field of QueueMember to Device using the hyphen separator
   same => n,Set(Device=${CUT(QueueMember,-,2)})

   ; Increase the value of the group inside the queue_members category by one
   same => n,Set(GROUP(queue_members)=${Technology}-${Device})

   ; Check if the group@category is greater than 1, and, if so, return Congestion()
   ; (too many channels)
   ;
   ; *** This line should not have any line breaks
   same => n,ExecIf($[${GROUP_COUNT(${Technology}-${Device}@queue_members)} > 1]
?Congestion())

   ; dial the agent
   same => n,Dial(${Technology}/${Device})
   same => n,Hangup()
1292281046|psy1-1292281041.87|7100|NONE|ENTERQUEUE||4165551212|1
1292281046|psy1-1292281041.87|7100|Local/9996@MemberConnector|RINGNOANSWER|0
1292281048|psy1-1292281041.87|7100|Local/9990@MemberConnector|CONNECT|2
|psy1-1292281046.90|0

1292284121|psy1-1292281041.87|7100|Local/9990@MemberConnector|COMPLETECALLER|2|3073|1
1292284222|MANAGER|7100|Local/9990@MemberConnector|REMOVEMEMBER|
1292284222|MANAGER|7200|Local/9990@MemberConnector|REMOVEMEMBER|
1292284491|MANAGER|7100|Local/9990@MemberConnector|ADDMEMBER|
1292284491|MANAGER|7200|Local/9990@MemberConnector|ADDMEMBER|
1292284519|psy1-1292284515.93|7100|NONE|ENTERQUEUE||4165551212|1
1292284519|psy1-1292284515.93|7100|Local/9996@MemberConnector|RINGNOANSWER|0
1292284521|psy1-1292284515.93|7100|Local/9990@MemberConnector|CONNECT|2
|psy1-1292284519.96|0

1292284552|MANAGER|7100|Local/9990@MemberConnector|REMOVEMEMBER|
1292284552|MANAGER|7200|Local/9990@MemberConnector|REMOVEMEMBER|
1292284562|psy1-1292284515.93|7100|Local/9990@MemberConnector|COMPLETECALLER|2|41|1

exten => 7012,1,Answer()

; *** This line should not have any line breaks
    same => n,Verbose(3,The state of SIP/0004F2060EB4 is 
${DEVICE_STATE(SIP/0004F2060EB4)})
    same => n,Hangup()
    -- The state of SIP/0004F2060EB4 is INUSE
[default]

exten => 1234,hint,SIP/phoneA&SIP/phoneB&SIP/phoneC

exten => 5555,hint,DAHDI/1

exten => 31337,hint,MeetMe:31337
[default]

exten => 1234,hint,SIP/phoneA&SIP/phoneB&SIP/phoneC
exten => 1234,1,Dial(SIP/phoneA&SIP/phoneB&SIP/phoneC)

exten => 5555,hint,DAHDI/1
exten => 5555,1,Dial(DAHDI/1)

exten => 31337,hint,MeetMe:31337
exten => 31337,1,MeetMe(31337,dM)
[phones]

exten => 7001,hint,SIP/0004F2060EB4
*CLI> core show hints

    -= Registered Asterisk Dial Plan Hints =-
           7001@phones   : SIP/0004F2060EB4    State:InUse    Watchers  0
----------------
- 1 hints registered
exten => 7013,1,Answer()
    same => n,Verbose(3,The state of 7001@phones is ${EXTENSION_STATE(7001@phones)})
    same => n,Hangup()
    -- The state of 7001@phones is INUSE
exten => example,1,Set(DEVICE_STATE(Custom:example)=BUSY)
exten => Verbose(1,The state of Custom:example is ${DEVICE_STATE(Custom:example)})
exten => example,hint,Custom:example
exten => DND_7015,hint,Custom:DND_7015
exten => DND_7015,1,Answer()
    same => n,GotoIf($["${DEVICE_STATE(Custom:DND_7015)}"="BUSY"]?turn_off:turn_on)

    same => n(turn_off),Set(DEVICE_STATE(Custom:DND_7015)=NOT_INUSE)
    same => n,Hangup()

    same => n(turn_on),Set(DEVICE_STATE(Custom:DND_7015)=BUSY)
    same => n,Hangup()
exten => 7015,1,GotoIf($["${DEVICE_STATE(Custom:DND_7015)}"="BUSY"]?busy:available)
    same => n(available),Verbose(3,DND is currently off for 7015.)
    same => n,Dial(SIP/exampledevice)
    same => n,Hangup()

    same => n(busy),Verbose(3,DND is on for 7015.)
    same => n,Playback(vm-theperson)
    same => n,Playback(digits/7&digits/0&digits/1&digits/5)
    same => n,Playback(vm-isunavail)
    same => n,Playback(vm-goodbye)
    same => n,Hangup()
;
; A hint so a phone can use BLF to signal the DND state.
;
exten => DND_7015,hint,Custom:DND_7015

;
; An extension to dial when the user presses the custom DND
; key on his phone.  This will toggle the state and will result
; in the light on the phone turning on or off.
;
exten => DND_7015,1,Answer()
    same => n,GotoIf($["${DEVICE_STATE(Custom:DND_7015)}"="BUSY"]?turn_off:turn_on)

    same => n(turn_off),Set(DEVICE_STATE(Custom:DND_7015)=NOT_INUSE)
    same => n,Hangup()

    same => n(turn_on),Set(DEVICE_STATE(Custom:DND_7015)=BUSY)
    same => n,Hangup()

;
; Example usage of the DND state.
;
exten => 7015,1,GotoIf($["${DEVICE_STATE(Custom:DND_7015)}"="BUSY"]?busy:available)
    same => n(available),Verbose(3,DND is currently off for 7015.)
    same => n,Dial(SIP/exampledevice)
    same => n,Hangup()

    same => n(busy),Verbose(3,DND is on for 7015.)
    same => n,Playback(vm-theperson)
    same => n,Playback(digits/7&digits/0&digits/1&digits/5)
    same => n,Playback(vm-isunavail)
    same => n,Playback(vm-goodbye)
    same => n,Hangup()
$ tar xvzf corosync-1.2.8.tar.gz
$ cd corosync-1.2.8
$ ./configure
$ make
$ sudo make install

$ tar xvzf openais-1.1.4.tar.gz
$ cd openais-1.1.4
$ ./configure
$ make
$ sudo make install
$ cd /path/to/asterisk
$ ./configure
$ make menuselect
$ make
$ sudo make install
$ sudo mkdir -p /etc/ais
$ cd openais-1.1.4
$ sudo cp conf/openais.conf.sample /etc/ais/openais.conf

$ sudo mkdir -p /etc/corosync
$ cd corosync-1.2.8
$ sudo cp conf/corosync.conf.sample /etc/corosync/corosync.conf
totem {
    ...
    interface {
        ringnumber: 0
        bindnetaddr: 10.24.22.144
        mcastaddr: 226.94.1.1
        mcastport: 5405
    }
}
$ man openais.conf
$ man corosync.conf
$ sudo aisexec -f
Nov 13 06:55:30 corosync [CLM   ] CLM CONFIGURATION CHANGE
Nov 13 06:55:30 corosync [CLM   ] New Configuration:
Nov 13 06:55:30 corosync [CLM   ]       r(0) ip(10.24.22.144) 
Nov 13 06:55:30 corosync [CLM   ]       r(0) ip(10.24.22.242) 
Nov 13 06:55:30 corosync [CLM   ] Members Left:
Nov 13 06:55:30 corosync [CLM   ] Members Joined:
Nov 13 06:55:30 corosync [CLM   ]       r(0) ip(10.24.22.242) 
Nov 13 06:55:30 corosync [TOTEM ] A processor joined or left the membership and a new 
membership was formed.
Nov 13 06:55:30 corosync [MAIN  ] Completed service synchronization, ready to provide
service.
[device_state]

type = event_channel
publish_event = device_state
subscribe_event = device_state
*CLI> ais evt show event channels 

=============================================================
=== Event Channels ==========================================
=============================================================
===
=== ---------------------------------------------------------
=== Event Channel Name: device_state
=== ==> Publishing Event Type: device_state
=== ==> Subscribing to Event Type: device_state
=== ---------------------------------------------------------
===
=============================================================
*CLI> ais clm show members 

=============================================================
=== Cluster Members =========================================
=============================================================
===
=== ---------------------------------------------------------
=== Node Name: 10.24.22.144
=== ==> ID: 0x9016180a
=== ==> Address: 10.24.22.144
=== ==> Member: Yes
=== ---------------------------------------------------------
===
=== ---------------------------------------------------------
=== Node Name: 10.24.22.242
=== ==> ID: 0xf216180a
=== ==> Address: 10.24.22.242
=== ==> Member: Yes
=== ---------------------------------------------------------
===
=============================================================
[devstate_test]

exten => foo,hint,Custom:abc
pbx1*CLI> dialplan set global DEVICE_STATE(Custom:abc) INUSE

    -- Global variable 'DEVICE_STATE(Custom:abc)' set to 'INUSE'
*CLI> core show hints

-= Registered Asterisk Dial Plan Hints =-
   foo@devstatetest        : Custom:abc        State:InUse       Watchers  0
*CLI> core set debug 1
*CLI> dialplan set global DEVICE_STATE(Custom:abc) NOT_INUSE

    -- Global variable 'DEVICE_STATE(Custom:abc)' set to 'NOT_INUSE'

[Nov 13 13:27:12] DEBUG[14801]: devicestate.c:652 
handle_devstate_change: Processing device state change for 'Custom:abc'
[Nov 13 13:27:12] DEBUG[14801]: devicestate.c:602 
process_collection: Adding per-server state of 'Not in use' for 'Custom:abc'
[Nov 13 13:27:12] DEBUG[14801]: devicestate.c:602 
process_collection: Adding per-server state of 'Not in use' for 'Custom:abc'
[Nov 13 13:27:12] DEBUG[14801]: devicestate.c:609 
process_collection: Aggregate devstate result is 'Not in use' for 'Custom:abc'
[Nov 13 13:27:12] DEBUG[14801]: devicestate.c:631 
process_collection: Aggregate state for device 'Custom:abc' has changed to 
'Not in use'

*CLI> dialplan set global DEVICE_STATE(Custom:abc) INUSE

    -- Global variable 'DEVICE_STATE(Custom:abc)' set to 'INUSE'

[Nov 13 13:29:30] DEBUG[14801]: devicestate.c:652 handle_devstate_change: 
Processing device state change for 'Custom:abc'
[Nov 13 13:29:30] DEBUG[14801]: devicestate.c:602 process_collection: 
Adding per-server state of 'Not in use' for 'Custom:abc'
[Nov 13 13:29:30] DEBUG[14801]: devicestate.c:602 process_collection: 
Adding per-server state of 'In use' for 'Custom:abc'
[Nov 13 13:29:30] DEBUG[14801]: devicestate.c:609 process_collection: 
Aggregate devstate result is 'In use' for 'Custom:abc'
[Nov 13 13:29:30] DEBUG[14801]: devicestate.c:631 process_collection: 
Aggregate state for device 'Custom:abc' has changed to 'In use'
*CLI> module show like jabber

Module                         Description                              Use Count 
res_jabber.so                  AJI - Asterisk Jabber Interface          0         
1 modules loaded
$ ls -l /usr/lib/asterisk/modules/res_jabber.so

-rwxr-xr-x 1 root root 837436 2010-11-12 15:33 /usr/lib/asterisk/modules/res_jabber.so
$ cd /path/to/asterisk
$ ./configure
$ make menuselect
$ make
$ sudo make install
[general]
autoregister = yes

[asterisk]
type = client
serverhost = jabber.shifteight.org
pubsub_node = pubsub.jabber.shifteight.org
username = server1@jabber.shifteight.org/astvoip1
secret = mypassword
distribute_events = yes
status = available
usetls = no
usesasl = yes
buddy = server2@jabber.shifteight.org/astvoip2
[general]
autoregister = yes

[asterisk]
type = client
serverhost = jabber.shifteight.org
pubsub_node = pubsub.jabber.shifteight.org
username = server2@jabber.shifteight.org/astvoip2
secret = mypassword
distribute_events = yes
status = available
usetls = no
usesasl = yes
buddy = server1@jabber.shifteight.org/astvoip1
*CLI> jabber show connected

Jabber Users and their status:
       User: server1@jabber.shifteight.org/astvoip1     - Connected
----
   Number of users: 1
*CLI> jabber show connected

Jabber Users and their status:
       User: server2@jabber.shifteight.org/astvoip2     - Connected
----
   Number of users: 1
*CLI> jabber show buddies

Jabber buddy lists
Client: server1@jabber.shifteight.org/astvoip1
    Buddy: server2@jabber.shifteight.org
           Resource: None
    Buddy: server2@jabber.shifteight.org/astvoip2
           Resource: None
*CLI> jabber show buddies

Jabber buddy lists
Client: server2@jabber.shifteight.org/astvoip2
    Buddy: server1@jabber.shifteight.org
           Resource: astvoip1
               node: http://www.asterisk.org/xmpp/client/caps
               version: asterisk-xmpp
               Jingle capable: yes
           Status: 1
           Priority: 0
    Buddy: server1@jabber.shifteight.org/astvoip1
           Resource: None
pbx*CLI> module show like app_meetme.so

Module                         Description                              Use Count 
0 modules loaded
*CLI> module show like app_meetme.so

Module                         Description                              Use Count 
app_meetme.so                  MeetMe conference bridge                 0         
1 modules loaded
*CLI> core show applications like SLA

    -= Matching Asterisk Applications =-
            SLAStation: Shared Line Appearance Station. 
              SLATrunk: Shared Line Appearance Trunk. 
    -= 2 Applications Matching =-
[line1]
type = trunk
device = DAHDI/1

[line2]
type = trunk
device = DAHDI/2
[station1]
type = station
device = SIP/station1
trunk = line1
trunk = line2

[station2]
type = station
device = SIP/station2
trunk = line1
trunk = line2

[station3]
type = station
device = SIP/station3
trunk = line1
trunk = line2

[station4]
type = station
device = SIP/station4
trunk = line1
trunk = line2
[station](!)
type = trunk
trunk = line1
trunk = line2

[station1](station)
device = SIP/station1

[station2](station)
device = SIP/station2

[station3](station)
device = SIP/station3

[station4](station)
device = SIP/station4
[line1]

exten => s,1,SLATrunk(line1)

[line2]

exten => s,1,SLATrunk(line2)
[sla_stations]

exten => station1,1,SLAStation(station1)
exten => station1_line1,hint,SLA:station1_line1
exten => station1_line1,1,SLAStation(station1_line1)
exten => station1_line2,hint,SLA:station1_line2
exten => station1_line2,1,SLAStation(station1_line2)

exten => station2,1,SLAStation(station2)
exten => station2_line1,hint,SLA:station2_line1
exten => station2_line1,1,SLAStation(station2_line1)
exten => station2_line2,hint,SLA:station2_line2
exten => station2_line2,1,SLAStation(station2_line2)

exten => station3,1,SLAStation(station3)
exten => station3_line1,hint,SLA:station3_line1
exten => station3_line1,1,SLAStation(station3_line1)
exten => station3_line2,hint,SLA:station3_line2
exten => station3_line2,1,SLAStation(station3_line2)

exten => station4,1,SLAStation(station4)
exten => station4_line1,hint,SLA:station4_line1
exten => station4_line1,1,SLAStation(station4_line1)
exten => station4_line2,hint,SLA:station4_line2
exten => station4_line2,1,SLAStation(station4_line2)
[line1]
type = trunk
device = Local/disa@line1_outbound

[line2]
type = trunk
device = Local/disa@line2_outbound
[station](!)
type = trunk
trunk = line1
trunk = line2

[station1](station)
device = SIP/station1

[station2](station)
device = SIP/station2

[station3](station)
device = SIP/station3

[station4](station)
device = SIP/station4
[line1]

exten => s,1,SLATrunk(line1)

;
; If the provider specifies your phone number when sending you
; a call, you will need another rule in the dialplan to match that.
;
exten => _X.,1,Goto(s,1)

[line2]

exten => s,1,SLATrunk(line2)

exten => _X.,1,Goto(s,1)
[sla_stations]

exten => station1,1,SLAStation(station1)
exten => station1_line1,hint,SLA:station1_line1
exten => station1_line1,1,SLAStation(station1_line1)
exten => station1_line2,hint,SLA:station1_line2
exten => station1_line2,1,SLAStation(station1_line2)

exten => station2,1,SLAStation(station2)
exten => station2_line1,hint,SLA:station2_line1
exten => station2_line1,1,SLAStation(station2_line1)
exten => station2_line2,hint,SLA:station2_line2
exten => station2_line2,1,SLAStation(station2_line2)

exten => station3,1,SLAStation(station3)
exten => station3_line1,hint,SLA:station3_line1
exten => station3_line1,1,SLAStation(station3_line1)
exten => station3_line2,hint,SLA:station3_line2
exten => station3_line2,1,SLAStation(station3_line2)

exten => station4,1,SLAStation(station4)
exten => station4_line1,hint,SLA:station4_line1
exten => station4_line1,1,SLAStation(station4_line1)
exten => station4_line2,hint,SLA:station4_line2
exten => station4_line2,1,SLAStation(station4_line2)
[line1_outbound]

exten => disa,1,DISA(no-password,line1_outbound)

;
; Add extensions for whatever numbers you would like to
; allow to be dialed.
;
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider)

[line2_outbound]

exten => disa,1,DISA(no-password,line2_outbound)

exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider)
[5001]
type = trunk
device = Local/disa@5001_outbound

[5001_phone1]
device = SIP/5001_phone1
trunk = 5001

[5001_phone2]
device = SIP/5001_phone2
trunk = 5001
exten => 5001,1,SLATrunk(5001)
[5001]

;
; This extension is needed if you want the shared extension to
; be used by default. In that case, have this extension dialed
; when the phone goes off-hook.
;
exten => 5001_phone1,1,SLAStation(5001_phone1)
;
; This is the extension that should be dialed when the 5001 key is
; pressed on 5001_phone1.
;
exten => 5001_phone1_5001,hint,SLA:5001_phone1_5001
exten => 5001_phone1_5001,1,SLAStation(5001_phone1_5001)

exten => 5001_phone2,1,SLAStation(5001_phone2)
exten => 5001_phone2_5001,hint,SLA:5001_phone2_5001
exten => 5001_phone2_5001,1,SLAStation(5001_phone2_5001)
[5001_outbound]

exten => disa,1,DISA(no-password,5001_outbound)

;
; This context will also need to be able to see whatever
; extensions you would like to be reachable from this extension.
;
include => pbx_extensions
[context_for_my_handset]
exten => 101,1,Playback(vm-intro)
exten => 101,n,Record(maingreeting.wav)
exten => 101,n,Wait(2)
exten => 101,n,Playback(maingreeting)
exten => 101,n,Hangup
[main_menu]

exten => s,1,Verbose(1, Caller ${CALLERID(all)} has entered the auto attendant)
    same => n,Answer()

; this sets the inter-digit timer
    same => n,Set(TIMEOUT(digit)=2)

; wait one second to establish audio
    same => n,Wait(1)

; If Mon-Fri 9-5 goto label daygreeting
    same => n,GotoIfTime(9:00-17:00,mon-fri,*,*?daygreeting:afterhoursgreeting)

    same => n(afterhoursgreeting),Background(after-hours) ; AFTER HOURS GREETING
    same => n,Goto(menuprompt)

    same => n(daygreeting),Background(daytime)   ; DAY GREETING
    same => n,Goto(menuprompt)

    same => n(menuprompt),Background(main-menu) ; MAIN MENU PROMPT
    same => n,WaitExten(4)                      ; more than 4 seconds is probably
                                                ; too much
    same => n,Goto(0,1)                         ; Treat as if caller has pressed '0'

exten => 1,1,Verbose(1, Caller ${CALLERID(all)} has entered the sales queue)
    same => n,Goto(Queues,7002,1)     ; Sales Queue - see Chapter 13 for details

exten => 2,1,Verbose(1, Caller ${CALLERID(all)} has entered the service queue)
    same => n,Goto(Queues,7001,1)     ; Service Queue - see Chapter 13 for details

exten => 3,1,Verbose(1, Caller ${CALLERID(all)} has requested address and fax info)
    same => n,Background()            ; Address and fax info
    same => n,Goto(s,menuprompt)      ; Take caller back to main menu prompt

exten => #,1,Verbose(1, Caller ${CALLERID(all)} is entering the directory)
    same => n,Directory(default)   ; Send the caller to the directory.
                                   ; Use InternalSets as the dialing context

exten => 0,1,Verbose(1, Caller ${CALLERID(all)} is calling the operator)
    same => n,Dial(SIP/operator)      ; Operator extension/queue

exten => i,1,Verbose(1, Caller ${CALLERID(all)} has entered an invalid selection)
    same => n,Playback(invalid)
    same => n,Goto(s,menuprompt)

exten => t,1,Verbose(1, Caller ${CALLERID(all)} has timed out)
    same => n,Goto(0,1)

; You will want to have a pattern match for the various extensions
; that you'll allow external callers to dial
; BUT DON'T JUST INCLUDE THE LocalSets CONTEXT
; OR EXTERNAL CALLERS WILL BE ABLE TO MAKE CALLS OUT OF YOUR SYSTEM

; WHATEVER YOU DO HERE, TEST IT CAREFULLY TO ENSURE EXTERNAL CALLERS
; WILL NOT BE ABLE TO DO ANYTHING BUT DIAL INTERNAL EXTENSIONS

exten => _1XX,1,Verbose(1,Call to an extension starting with '1'
    same => n,Goto(InternalSets,${EXTEN},1)

[from-pstn] ; an analog line that has context=from-pstn (typically a DAHDI channel)
exten => s,1,Goto(main_menu,s,1)
[incoming] ; a DID coming in on a channel with context=incoming (PRI, SIP, or IAX)
exten => 4169671111,1,Goto(main_menu,s,1)

$ sudo yum install -y postgresql-server
Install       3 Package(s)
Upgrade       0 Package(s)

Total download size: 6.9 M
Is this ok [y/N]: y
$ sudo service postgresql start
$ sudo apt-get install postgresql
...
After this operation, 19.1MB of additional disk space will be used.
Do you want to continue [Y/n]? y
$ sudo yum install mysql-server
Install       5 Package(s)
Upgrade       0 Package(s)

Total download size: 27 M
Is this ok [y/N]: y
$ sudo service mysqld start
$ sudo apt-get install mysql-server
Need to get 24.0MB of archives.
After this operation, 60.6MB of additional disk space will be used.
Do you want to continue [Y/n]? y
$ sudo su - postgres
$ createuser -P
Enter name of user to add: asterisk
Enter password for new user:
Enter it again:
Shall the new role be a superuser? (y/n) n
Shall the new user be allowed to create databases? (y/n) y
Shall the new user be allowed to create more new users? (y/n) n
CREATE ROLE
# TYPE DATABASE USER CIDR-ADDRESS METHOD
# TYPE  DATABASE USER     CIDR-ADDRESS  METHOD
host    all      asterisk 127.0.0.1/32  md5
local   all      asterisk               trust
host    all     asterisk          ::1/128                               md5
[28000][unixODBC]FATAL:  no pg_hba.conf entry for host "::1", user "asterisk", 
database "asterisk", SSL off
[ISQL]ERROR: Could not SQLConnect
$ createdb --owner=asterisk asterisk
CREATE DATABASE
$ psql -d template1
template1=# "ALTER USER asterisk WITH PASSWORD 'password'"
template1=# \q
$ exit
$ sudo service postgresql restart
$ sudo /etc/init.d/postgresql-8.4 restart
$ psql -h 127.0.0.1 -U asterisk 
Password for user asterisk:

Welcome to psql 8.1.21, the PostgreSQL interactive terminal.

Type:  \copyright for distribution terms
       \h for help with SQL commands
       \? for help with psql commands
       \g or terminate with semicolon to execute query
       \q to quit

asterisk=>
$ sudo /usr/bin/mysql_secure_installation
$ mysql -u root -p
Enter password:
mysql> CREATE USER 'asterisk'@'%' IDENTIFIED BY 'some_secret_password';
Query OK, 0 rows affected (0.00 sec)
mysql> CREATE DATABASE asterisk;
Query OK, 1 rows affected (0.00 sec)
mysql> GRANT ALL PRIVILEGES ON asterisk.* TO 'asterisk'@'%';
Query OK, 0 rows affected (0.00 sec)
mysql> exit
Bye
# mysql -u asterisk -p asterisk
Enter password:

mysql>
$ sudo yum install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel
$ sudo apt-get install unixODBC unixODBC-dev
$ sudo yum install mysql-connector-odbc
$ sudo apt-get install odbc-postgresql
$ sudo apt-get install libmyodbc
[PostgreSQL]
Description     = ODBC for PostgreSQL
Driver          = /usr/lib/libodbcpsql.so
Setup           = /usr/lib/libodbcpsqlS.so
FileUsage       = 1
[PostgreSQL]
Description     = ODBC for PostgreSQL
Driver          = /usr/lib/odbc/psqlodbca.so
Setup           = /usr/lib/odbc/libodbcpsqlS.so
FileUsage       = 1
$ odbcinst -q -d
[PostgreSQL]
[asterisk-connector]
Description         = PostgreSQL connection to 'asterisk' database
Driver              = PostgreSQL
Database            = asterisk
Servername          = localhost
UserName            = asterisk
Password            = welcome
Port                = 5432
Protocol            = 8.1
ReadOnly            = No
RowVersioning       = No
ShowSystemTables    = No
ShowOidColumn       = No
FakeOidIndex        = No
ConnSettings        =

[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/libmyodbc3.so
Setup = /usr/lib/libodbcmyS.so
FileUsage = 1
[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/odbc/libmyodbc.so
Setup = /usr/lib/odbc/libodbcmyS.so
FileUsage = 1
# odbcinst -q -d
[MySQL]
[asterisk-connector]
Description           = MySQL connection to 'asterisk' database
Driver                = MySQL
Database              = asterisk
Server                = localhost
UserName              = asterisk
Password              = welcome
Port                  = 3306
Socket                = /var/lib/mysql/mysql.sock
$ sudo yum install freetds
$ sudo apt-get install freetds
$ sudo cat > /etc/odbcinst.ini
[FreeTDS]
Description = ODBC for Microsoft SQL
Driver      = /usr/lib/libtdsodbc.so
UsageCount  = 1
Threading   = 2
Ctrl+D
$ odbcinst -q -d
[FreeTDS]
[asterisk-connector]
Description     = MS SQL connection to 'asterisk' database
Driver          = FreeTDS
Database        = asterisk
Server          = 192.168.100.1
UserName        = asterisk
Password        = welcome
Trace           = No
TDS_Version     = 7.0
Port            = 1433
$ echo "select 1" | isql -v asterisk-connector
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL> +------------+
| ?column?   |
+------------+
| 1          |
+------------+
SQLRowCount returns 1
1 rows fetched
$ exit
$ cd ~/src/asterisk-complete/asterisk/1.8
$ ./configure
$ make menuselect
$ make install
[asterisk]
enabled => yes
dsn => asterisk-connector
username => asterisk
password => welcome
pooling => no
limit => 1
pre-connect => yes
*CLI> odbc show

ODBC DSN Settings
-----------------

  Name:   asterisk
  DSN:    asterisk-connector

    Last connection attempt: 1969-12-31 19:00:00
  Pooled: No
  Connected: Yes
[span3pri]
exten => _50054XX,1,NoOp()
   same => n,Set(CDR(accountcode)=pricall)
   same => n,GotoIf($[${ODBC_ANIBLOCK(${CALLERID(number)})}]?busy)
   same => n(dial),Dial(DAHDI/G1/${EXTEN})
   same => n(busy),Busy(10)
   same => n,Hangup

[ANIBLOCK]
dsn=telesys
readsql=SELECT IF(COUNT(1)>0, 1, 0) FROM Aniblock WHERE NUMBER='${ARG1}'
; sip.conf
; HOT DESK USERS
[0000FFFF0001]
type=friend
host=dynamic
secret=my_special_secret
context=hotdesk
qualify=yes

[0000FFFF0002]
type=friend
host=dynamic
secret=my_special_secret
context=hotdesk
qualify=yes

; END HOT DESK USERS
$ sudo su - postgres
$ psql -U asterisk -h localhost asterisk
Password:
CREATE TABLE ast_hotdesk
(
  id serial NOT NULL,
  extension int8,
  first_name text,
  last_name text,
  cid_name text,
  cid_number varchar(10),
  pin int4,
  context text,
  status bool DEFAULT false,
  "location" text,
  CONSTRAINT ast_hotdesk_id_pk PRIMARY KEY (id)
)
WITHOUT OIDS;
$ mysql -u asterisk -p asterisk
Enter password:
CREATE TABLE ast_hotdesk
(
  id serial NOT NULL,
  extension int8,
  first_name text,
  last_name text,
  cid_name text,
  cid_number varchar(10),
  pin int4,
  context text,
  status bool DEFAULT false,
  location text,
  CONSTRAINT ast_hotdesk_id_pk PRIMARY KEY (id)
);
asterisk=> INSERT INTO ast_hotdesk ('extension', 'first_name', 'last_name',\
'cid_name','cid_number', 'pin', 'context', 'location') \
VALUES (1101, 'Leif', 'Madsen', 'Leif Madsen', '4165551101', '555',\
'longdistance','0000FFFF0001');
mysql> INSERT INTO ast_hotdesk (extension, first_name, last_name, cid_name, 
cid_number, pin, context, location)
VALUES (1101, 'Leif', 'Madsen', 'Leif Madsen', 
'4165551101', '555', 'longdistance', '0000FFFF0001');
mysql> SELECT * FROM ast_hotdesk;
| id | extension | first_name | last_name      | cid_name           | cid_number   
|----+-----------+------------+----------------+--------------------+--------------
|  1 | 1101      | "Leif"     | "Madsen"       | "Leif Madsen"      | "4165551101" 
|  2 | 1102      | "Jim"      | "Van Meggelen" | "Jim Van Meggelen" | "4165551102" 
|  3 | 1103      | "Russell"  | "Bryant"       | "Russell Bryant"   | "4165551103" 
|  4 | 1104      | "Mark"     | "Spencer"      | "Mark Spencer"     | "4165551104" 
|  5 | 1105      | "Kevin"    | "Fleming"      | "Kevin Fleming"    | "4165551105" 

|  pin  | context         | status  | location       |$
+-------+-----------------+---------+----------------+
| "555" | "longdistance"  | "TRUE"  | "0000FFFF0001" |
| "556" | "longdistance"  | "FALSE" | ""             |
| "557" | "local"         | "FALSE" | ""             |
| "558" | "international" | "FALSE" | ""             |
| "559" | "local"         | "FALSE" | ""             |
; extensions.conf
; Hot-Desking Feature
[hotdesk]
; Hot Desk Login
exten => _#110[1-5],1,NoOp()
   same => n,Set(E=${EXTEN:1})  ; strip off the leading hash (#) symbol
   same => n,Verbose(1,Hot Desk Extension ${E} is changing status)
   same => n,Verbose(1,Checking current status of extension ${E})
   same => n,Set(${E}_STATUS=${HOTDESK_INFO(status,${E})})
   same => n,Set(${E}_PIN=${HOTDESK_INFO(pin,${E})})
[INFO]
prefix=HOTDESK
dsn=asterisk
readsql=SELECT ${ARG1} FROM ast_hotdesk WHERE extension = '${ARG2}'
exten => s,n,Set(RETURNED_VALUE=${HOTDESK_INFO(status,1101)})
[ALL_AVAIL_EXTENS]
prefix=GET
dsn=asterisk-connector
mode=multirow
readsql=SELECT extension FROM ast_hotdesk WHERE status = '${ARG1}'
[multirow_example]
exten => start,1,Verbose(1,Looping example)
   same => n,Set(ODBC_ID=${GET_ALL_AVAIL_EXTENS(1)})
   same => n,GotoIf($[${ODBCROWS} < 1]?no_rows,1)
   same => n,Set(COUNTER=1)
   same => n,While($[${COUNTER} <= ${ODBCROWS}])
   same => n,Set(AVAIL_EXTEN_${COUNTER}=${ODBC_FETCH(${ODBC_ID})})
   same => n,Set(COUNTER=$[${COUNTER + 1])
   same => n,EndWhile()
   same => n,ODBCFinish()

exten => no_rows,1,Verbose(1,No rows returned)
   same => n,Playback(silence/1&invalid)
   same => n,Hangup()
[multirow_example_2]
exten => start,1,Verbose(1,Looping example with break)
   same => n,Set(ODBC_ID=${GET_ALL_AVAIL_EXTENS(1)})
   same => n(loop_start),NoOp()
   same => n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
   same => n,GotoIf($["${ODBC_FETCH_RESULT}" = "FAILURE"]?cleanup,1)
   same => n,GotoIf($["${ROW_RESULT}" = "1104"]?good_exten,1)
   same => n,Goto(loop_start)

exten => cleanup,1,Verbose(1,Cleaning up after all iterations)
   same => n,Verbose(1,We did not find the extension we wanted)
   same => n,ODBCFinish(${ODBC_ID})
   same => n,Hangup()

exten => good_exten,1,Verbose(1,Extension we want is available)
   same => n,ODBCFinish(${ODBC_ID})
   same => n,Verbose(1,Perform some action we wanted)
   same => n,Hangup()
exten => _110[1-5],n,Set(${E}_STATUS=${HOTDESK_INFO(status,${E})})
   same => n,Set(${E}_PIN=${HOTDESK_INFO(pin,${E})})
readsql=SELECT pin,status FROM ast_hotdesk WHERE extension = '${E}'
exten => _110[1-5],n,Set(ARRAY(${E}_PIN,${E}_STATUS)=${HOTDESK_INFO(${E})})
   same => n,Set(${E}_STATUS=${HOTDESK_INFO(status,${E})})
   same => n,Set(${E}_PIN=${HOTDESK_INFO(pin,${E})})
   same => n,GotoIf($[${ODBCROWS} < 0]?invalid_user,1)
; check if ${E}_STATUS is NULL
   same => n,GotoIf($[${${E}_STATUS} = 1]?logout,1:login,1)
exten => login,1,NoOp() ; set initial counter values
   same => n,Set(PIN_TRIES=1)     ; pin tries counter 
   same => n,Set(MAX_PIN_TRIES=3) ; set max number of login attempts
   same => n,Playback(silence/1)  ; play back some silence so first prompt is 
                                  ; not cut off
   same => n(get_pin),NoOp()
   same => n,Set(PIN_TRIES=$[${PIN_TRIES} + 1])   ; increase pin try counter
   same => n,Read(PIN_ENTERED,enter-password,${LEN(${${E}_PIN})})
   same => n,GotoIf($["${PIN_ENTERED}" = "${${E}_PIN}"]?valid_login,1)
   same => n,Playback(pin-invalid)
   same => n,GotoIf($[${PIN_TRIES} <= ${MAX_PIN_TRIES}]?get_pin:login_fail,1)

exten => valid_login,1,NoOp()
; CUT off the channel technology and assign it to the LOCATION variable
   same => n,Set(LOCATION=${CUT(CHANNEL,/,2)})
; CUT off the unique identifier and save the remainder to the LOCATION variable
   same => n,Set(LOCATION=${CUT(LOCATION,-,1)})
; func_odbc.conf
[CHECK_PHONE_LOGINS]
prefix=HOTDESK
dsn=asterisk

; *** This line should have no line breaks
readsql=SELECT COUNT(status) FROM ast_hotdesk WHERE status = '1' AND 
location = '${ARG1}'
; Continuation of the valid_login extension below
   same => n,Set(USERS_LOGGED_IN=${HOTDESK_CHECK_PHONE_
LOGINS(${LOCATION})})
   same => n,GotoIf($[${USERS_LOGGED_IN} > 0]?logout_login,1)
   same => n(set_login_status),NoOp()

; Set the status for the phone to '1' and where the agent is logged into
   same => n,Set(HOTDESK_STATUS(${E})=1,${LOCATION})
   same => n,GotoIf($[${ODBCROWS} < 1]?error,1)
   same => n,Playback(agent-loginok)
   same => n,Hangup()
[STATUS]
prefix=HOTDESK
dsn=asterisk

; *** This line should have no line breaks
writesql=UPDATE ast_hotdesk SET status = '${VAL1}', 
location = '${VAL2}' WHERE extension = '${ARG1}'
Set(HOTDESK_STATUS(${E})=1,${LOCATION})
   same => n,Set(LOGGED_IN_ID=${HOTDESK_LOGGED_IN_USER(${LOCATION})})
exten => logout_login,1,NoOp()
; set all logged-in users on this device to logged-out status
   same => n,Set(LOGGED_IN_ID=${HOTDESK_LOGGED_IN_USER(${LOCATION})})
   same => n(start_loop),NoOp()
   same => n,Set(WHO=${ODBC_FETCH(${LOGGED_IN_ID})})
   same => n,GotoIf($["${ODBC_FETCH_STATUS}" = "FAILURE"]?cleanup)
   same => n,Set(HOTDESK_STATUS(${WHO})=0)           ; log out phone
   same => n,Goto(start_loop)
   same => n(cleanup),ODBCFinish(${LOGGED_IN_ID})
   same => n,Goto(valid_login,set_login_status)      ; return to logging in

exten => logout,1,NoOp()
   same => n,Set(HOTDESK_STATUS(${E})=0)
   same => n,GotoIf($[${ODBCROWS} < 1]?error,1)
   same => n,Playback(silence/1&agent-loggedoff)
   same => n,Hangup()

exten => login_fail,1,NoOp()
   same => n,Playback(silence/1&login-fail)
   same => n,Hangup()

exten => error,1,NoOp()
   same => n,Playback(silence/1&connection-failed)
   same => n,Hangup()

exten => invalid_user,1,NoOp()
   same => n,Verbose(1,Hot Desk extension ${E} does not exist)
   same => n,Playback(silence/2&invalid)
   same => n,Hangup()
include => hotdesk_outbound
[PHONE_STATUS]
prefix=HOTDESK
dsn=asterisk
readsql=SELECT extension FROM ast_hotdesk WHERE status = '1'
readsql+= AND location = '${ARG1}'
[hotdesk_outbound]
exten => _X.,1,NoOp()
   same => n,Set(LOCATION=${CUT(CHANNEL,/,2)})
   same => n,Set(LOCATION=${CUT(LOCATION,-,1)})
   same => n,Set(WHO=${HOTDESK_PHONE_STATUS(${LOCATION})})
   same => n,GotoIf($[${ISNULL(${WHO})}]?no_outgoing,1)
   same => n,Set(${WHO}_CID_NAME=${HOTDESK_INFO(cid_name,${WHO})})
   same => n,Set(${WHO}_CID_NUMBER=${HOTDESK_INFO(cid_number,${WHO})})
   same => n,Set(${WHO}_CONTEXT=${HOTDESK_INFO(context,${WHO})})
   same => n,Goto(${${WHO}_CONTEXT},${EXTEN},1)

[international]
exten => _011.,1,NoOp()
   same => n,Set(E=${EXTEN})
   same => n,Goto(outgoing,call,1)

exten => i,1,NoOp()
   same => n,Playback(silence/2&sorry-cant-let-you-do-that2)
   same => n,Hangup()

include => longdistance

[longdistance]
exten => _1NXXNXXXXXX,1,NoOp()
   same => n,Set(E=${EXTEN})
   same => n,Goto(outgoing,call,1)

exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)

exten => i,1,NoOp()
   same => n,Playback(silence/2&sorry-cant-let-you-do-that2)
   same => n,Hangup()

include => local

[local]
exten => _416NXXXXXX,1,NoOp()
   same => n,Set(E=${EXTEN})
   same => n,Goto(outgoing,call,1)

exten => i,1,NoOp()
   same => n,Playback(silence/2&sorry-cant-let-you-do-that2)
   same => n,Hangup()
[outgoing]
exten => call,1,NoOp()
   same => n,Set(CALLERID(name)=${${WHO}_CID_NAME})
   same => n,Set(CALLERID(number)=${${WHO}_CID_NUMBER})
   same => n,Dial(SIP/service_provider/${E})
   same => n,Playback(silence/2&pls-try-call-later)
   same => n,Hangup()
[service_provider]
type=friend
host=switch1.service_provider.net
defaultuser=my_username
fromuser=my_username
secret=welcome
context=incoming
canreinvite=no
disallow=all
allow=ulaw
; /etc/asterisk/extconfig.conf
filename.conf => driver,database[,table]
$ cd /etc/asterisk
$ mv musiconhold.conf musiconhold.conf.old
*CLI> core restart now
*CLI> moh show classes
*CLI>
> INSERT INTO ast_config (filename,category,var_name,var_val)
  VALUES ('musiconhold.conf','default','mode','files');

> INSERT INTO ast_config (filename,category,var_name,var_val)
  VALUES ('musiconhold.conf','default','directory','/var/lib/asterisk/moh');
asterisk=# SELECT filename,category,var_name,var_val FROM ast_config;

 filename         | category       | var_name     | var_val
------------------+----------------+--------------+------------------------
 musiconhold.conf | default        | mode         | files
 musiconhold.conf | default        | directory    | /var/lib/asterisk/moh
(2 rows)
musiconhold.conf => odbc,asterisk,ast_config
*CLI> module reload res_musiconhold.so
*CLI> moh show classes
Class: general
        Mode: files
        Directory: /var/lib/asterisk/moh
; extconfig.conf
sippeers => driver,database[,table]
; extconfig.conf
sippeers => odbc,asterisk,ast_sipfriends
sipusers => odbc,asterisk,ast_sipfriends
WARNING: Realtime table ast_sipfriends@asterisk requires column 
'name', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'ipaddr', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'port', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'regseconds', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'defaultuser', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'fullcontact', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'regserver', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'useragent', but that column does not exist!

WARNING: Realtime table ast_sipfriends@asterisk requires column 
'lastms', but that column does not exist!

Name/username              Host         Dyn Port Status      Realtime
0000FFFF0008/0000FFFF0008  172.16.0.160  D  5060 Unmonitored Cached RT
+--------+--------------+---------+-----------+---------+--------------+
| type   | name         | secret  | context   | host    | ipaddr       |
+--------+--------------+---------+-----------+---------+--------------+
| friend | 0000FFFF0008 | welcome | LocalSets | dynamic | 172.16.0.160 | 
+--------+--------------+---------+-----------+---------+--------------+
+------+------------+--------------+-------------------------------------+
| port | regseconds | defaultuser  | fullcontact                         |
+------+------------+--------------+-------------------------------------+
| 5060 | 1283928895 | 0000FFFF0008 | sip:0000FFFF0008@172.16.0.160:52722 |
+------+------------+--------------+-------------------------------------+
+-----------+-----------------+--------+
| regserver | useragent       | lastms |
+-----------+-----------------+--------+
| NULL      | Zoiper rev.6739 |      0 | 
+-----------+-----------------+--------+
[options]
systemname=toronto
; cdr_adaptive_odbc.conf
[adaptive_connection]
connection=asterisk
table=cdr
*CLI> cdr show status

Call Detail Record (CDR) settings
----------------------------------
  Logging:                    Enabled
  Mode:                       Simple
  Log unanswered calls:       No

* Registered Backends
  -------------------
    cdr-syslog
    Adaptive ODBC
    cdr-custom
    csv
    cdr_manager
sql> ALTER TABLE cdr ADD COLUMN route_rate varchar(10);
*CLI> module reload cdr_adaptive_odbc.so
exten => _NXXNXXXXXX,1,Verbose(1,Example of adaptive ODBC usage)
   same => n,Set(CDR(route_rate)=0.01)
   same => n,Dial(SIP/my_itsp/${EXTEN})
   same => n,Hangup()
+--------------+----------+---------+------------+
| src          | duration | billsec | route_rate |
+--------------+----------+---------+------------+
| 0000FFFF0008 | 37       | 30      | 0.01       | 
+--------------+----------+---------+------------+

[mysql_connection]
connection=asterisk_mysql
table=cdr

[mssql_connection]
connection=production_mssql
table=call_records
[mssql_connection]
connection=production_mssql
table=call_records
alias src => Source
alias dst => Destination
alias accountcode => AccountCode
alias billsec => BillableTime
[logging_for_device_0000FFFF0008]
connection=asterisk_mysql
table=cdr_for_0000FFFF0008
filter src => 0000FFFF0008
[mysql_connection]
connection=asterisk_mysql
table=cdr

[filtered_mysql_connection]
connection=asterisk_mysql
table=cdr
filter src => 0000FFFF0008
static "DoNotCharge" => accountcode
$ psql -h localhost -U asterisk asterisk
Password:
CREATE FUNCTION loin (cstring) RETURNS lo AS 'oidin' LANGUAGE internal
IMMUTABLE STRICT;

CREATE FUNCTION loout (lo) RETURNS cstring AS 'oidout' LANGUAGE internal
IMMUTABLE STRICT;

CREATE FUNCTION lorecv (internal) RETURNS lo AS 'oidrecv' LANGUAGE internal
IMMUTABLE STRICT;

CREATE FUNCTION losend (lo) RETURNS bytea AS 'oidrecv' LANGUAGE internal
IMMUTABLE STRICT;

CREATE TYPE lo ( INPUT = loin, OUTPUT = loout, RECEIVE = lorecv, SEND = losend,
INTERNALLENGTH = 4, PASSEDBYVALUE );

CREATE CAST (lo AS oid) WITHOUT FUNCTION AS IMPLICIT;
CREATE CAST (oid AS lo) WITHOUT FUNCTION AS IMPLICIT;
CREATE FUNCTION vm_lo_cleanup() RETURNS "trigger"
    AS $$
    declare
      msgcount INTEGER;
    begin
      -- raise notice 'Starting lo_cleanup function for large object with oid
         %',old.recording;
      -- If it is an update action but the BLOB (lo) field was not changed,
         don't do anything
      if (TG_OP = 'UPDATE') then
        if ((old.recording = new.recording) or (old.recording is NULL)) then
          raise notice 'Not cleaning up the large object table,
          as recording has not changed';
          return new;
        end if;
      end if;
      if (old.recording IS NOT NULL) then
        SELECT INTO msgcount COUNT(*) AS COUNT FROM voicemessages WHERE recording
        = old.recording;
        if (msgcount > 0) then
          raise notice 'Not deleting record from the large object table, as object
          is still referenced';
          return new;
        else
          perform lo_unlink(old.recording);
          if found then
            raise notice 'Cleaning up the large object table';
            return new;
          else
            raise exception 'Failed to clean up the large object table';
            return old;
          end if;
        end if;
      else
        raise notice 'No need to clean up the large object table, 
        no recording on old row';

        return new;
      end if;
    end$$
    LANGUAGE plpgsql;
CREATE TABLE voicemessages
(
  uniqueid serial PRIMARY KEY,
  msgnum int4,
  dir varchar(80),
  context varchar(80),
  macrocontext varchar(80),
  callerid varchar(40),
  origtime varchar(40),
  duration varchar(20),
  mailboxuser varchar(80),
  mailboxcontext varchar(80),
  recording lo,
  label varchar(30),
  "read" bool DEFAULT false,
  flag varchar(10)
);
CREATE TRIGGER vm_cleanup AFTER DELETE OR UPDATE ON voicemessages FOR EACH ROW
EXECUTE PROCEDURE vm_lo_cleanup();
[general]
format=wav49
odbcstorage=asterisk
odbctable=voicemessages
[default]
1000 => 1000,J.P. Wiser
*CLI> module unload app_voicemail.so
  == Unregistered application 'VoiceMail'
  == Unregistered application 'VoiceMailMain'
  == Unregistered application 'MailboxExists'
  == Unregistered application 'VMAuthenticate'

*CLI> module load app_voicemail.so
 Loaded /usr/lib/asterisk/modules/app_voicemail.so => 
(Comedian Mail (Voicemail System))
  == Registered application 'VoiceMail'
  == Registered application 'VoiceMailMain'
  == Registered application 'MailboxExists'
  == Registered application 'VMAuthenticate'
  == Parsing '/etc/asterisk/voicemail.conf': Found
*CLI> voicemail show users for default
Context    Mbox  User                      Zone       NewMsg
default    1000  J.P. Wiser                                0
[odbc_vm_test]
exten => 100,1,VoiceMail(1000@default)      ; leave a voicemail
exten => 200,1,VoiceMailMain(1000@default)  ; retrieve a voicemail
*CLI> dialplan reload
[odbc_test_user]
type=friend
secret=supersecret
context=odbc_vm_test
host=dynamic
qualify=yes
disallow=all
allow=ulaw
allow=gsm
*CLI> module reload chan_sip.so
*CLI> sip show users like odbc_test_user
Username         Secret           Accountcode      Def.Context      ACL  NAT
odbc_test_user   supersecret                       odbc_vm_test     No   RFC3581
    -- Executing VoiceMail("SIP/odbc_test_user-10228cac", "1000@default") in new 
       stack
    -- Playing 'vm-intro' (language 'en')
    -- Playing 'beep' (language 'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/1000/tmp/dlZunm
       format: wav49, 0x101f6534
    -- User ended message by pressing #
    -- Playing 'auth-thankyou' (language 'en')
  == Parsing '/var/spool/asterisk/voicemail/default/1000/INBOX/msg0000.txt': Found
*CLI>
    -- Executing VoiceMailMain("SIP/odbc_test_user-10228cac", 
       "1000@default") in new stack
    -- Playing 'vm-password' (language 'en')
    -- Playing 'vm-youhave' (language 'en')
    -- Playing 'digits/1' (language 'en')
    -- Playing 'vm-INBOX' (language 'en')
    -- Playing 'vm-message' (language 'en')
    -- Playing 'vm-onefor' (language 'en')
    -- Playing 'vm-INBOX' (language 'en')
    -- Playing 'vm-messages' (language 'en')
    -- Playing 'vm-opts' (language 'en')
    -- Playing 'vm-first' (language 'en')
    -- Playing 'vm-message' (language 'en')
  == Parsing '/var/spool/asterisk/voicemail/default/1000/INBOX/msg0000.txt': Found
$ psql -h localhost -U asterisk asterisk
Password:
localhost=# SELECT uniqueid,dir,callerid,mailboxcontext,recording FROM voicemessages;
uniqueid | dir                                              | callerid     
---------+--------------------------------------------------+--------------
1        | /var/spool/asterisk/voicemail/default/1000/INBOX | +18005551212 

| mailboxcontext | recording |
+----------------+-----------+
| default        | 47395     |
(1 row)
localhost=# \lo_list
    Large objects
  ID   | Description
-------+-------------
 47395 |
(1 row)
localhost=# \lo_export 47395 /tmp/voicemail-47395.wav
lo_export
$ play /tmp/voicemail-47395.wav

Input Filename : /tmp/voicemail-47395.wav
Sample Size    : 8-bits
Sample Encoding: wav
Channels       : 1
Sample Rate    : 8000

Time: 00:06.22 [00:00.00] of 00:00.00 (  0.0%) Output Buffer: 298.36K

Done.
$ mysql -u asterisk -p asterisk
Enter password:
mysql> SELECT uniqueid, msgnum, callerid, mailboxuser, mailboxcontext, `read`
    -> FROM voicemessages;
+----------+--------+------------------------------+-------------
| uniqueid | msgnum | callerid                     | mailboxuser 
+----------+--------+------------------------------+-------------
|        1 |      0 | "Leif Madsen" <100>          | 100         
|        2 |      1 | "Leif Madsen" <100>          | 100         
|        3 |      2 | "Leif Madsen" <100>          | 100         
|        5 |      0 | "Julie Bryant" <12565551111> | 100         
+----------+--------+------------------------------+-------------
+----------------+------+
| mailboxcontext | read |
+----------------+------+
| shifteight.org |    0 |
| shifteight.org |    0 |
| shifteight.org |    0 |
| default        |    0 |
+----------------+------+
mysql> SELECT recording FROM voicemessages WHERE uniqueid = '5'
    -> DUMPFILE '/tmp/voicemail_recording.wav';
$ play /tmp/voicemail_recording.wav 

voicemail_recording.wav:

 File Size: 7.28k     Bit Rate: 13.1k
  Encoding: GSM           
  Channels: 1 @ 16-bit   
Samplerate: 8000Hz       
Replaygain: off         
  Duration: 00:00:04.44  

In:100%  00:00:04.44 [00:00:00.00] Out:35.5k [      |      ] Hd:4.4 Clip:0    
Done.
Read(variable[,filename[&filename2...]][,maxdigits][,option][,attempts][,timeout])
$ sudo yum -y install libcurl-devel
$ sudo apt-get install libcurl4-openssl-dev
exten => *764,1,Verbose(2, Run CURL to get IP address from whatismyip.org)
    same => n,Answer()
    same => n,Set(MyIPAddressIs=${CURL(http://www.whatismyip.org/)})
    same => n,SayAlpha(${MyIPAddressIs})
    same => n,Hangup()

[prompts]
exten => s,1,Answer
exten => s,n,Set(step1count=0) ; Initialize counters

; If we get no response after 3 times, we stop asking
   same => n(beginning),GotoIf($[${step1count} > 2]?end)
   same => n,Read(which,prompt-instructions,3)
   same => n,Set(step1count=$[${step1count} + 1])

; All prompts must be 3 digits in length
   same => n,GotoIf($[${LEN(${which})} < 3]?beginning) 
   same => n,Set(step1count=0) ; We have a successful response, so reset our counters
   same => n,Set(step2count=0)

   same => n(step2),Set(step2count=$[${step2count} + 1])
   same => n,GotoIf($[${step2count} > 2]?beginning) ; No response after 3 tries

; If the file doesn't exist, then don't ask whether to play it
   same => n,GotoIf($[${STAT(f,${which}.wav)} = 0]?recordonly) 
   same => n,Background(prompt-tolisten)

   same => n(recordonly),Background(prompt-torecord)
   same => n,WaitExten(10) ; Wait 10 seconds for a response
   same => n,Goto(step2)

exten => 1,1,Set(step2count=0)
   same => n,Background(${which})
   same => n,Goto(s,step2)

exten => 2,1,Set(step2count=0)
   same => n,Playback(prompt-waitforbeep)
   same => n,Record(${CHANNEL(uniqueid)}.wav)

   same => n(listen),Playback(${CHANNEL(uniqueid)})
   same => n,Set(step3count=0)
   same => n,Read(saveornot,prompt-1tolisten-2tosave-3todiscard,1)
   same => n,GotoIf($["${saveornot}" = "1"]?listen)
   same => n,GotoIf($["${saveornot}" = "2"]?saveit)
   same => n,System(rm -f /var/lib/asterisk/sounds/${CHANNEL(uniqueid)}.wav)
   same => n,Goto(s,beginning)

   same => n(saveit),System(mv -f ${CHANNEL(uniqueid)}.wav ${which}.wav)
   same => n,Playback(prompt-saved)
   same => n,Goto(s,beginning)
$ sudo yum install neon-devel
$ sudo yum --enablerepo=epel install libical-devel
$ sudo apt-get install libneon27-dev
$ sudo apt-get install libical-dev
$ cat >> calendar.conf
[myGoogleCal]
type=caldav
url=https://www.google.com/calendar/dav/<Gmail Email Address>/events/
user=<Gmail Email Address>
secret=<Gmail Password>
refresh=15
timeframe=60
Ctrl+D
$ asterisk -r
*CLI> module load res_calendar.so
*CLI> module reload
*CLI> calendar show calendars
Calendar             Type       Status
--------             ----       ------
myGoogleCal          caldav     busy
*CLI> calendar show calendar myGoogleCal
Name              : myGoogleCal         
Notify channel    :                     
Notify context    :                     
Notify extension  :                     
Notify applicatio :                     
Notify appdata    :                     
Refresh time      : 15
Timeframe         : 60
Autoreminder      : 0
Events
------
Summary     : Awesome Call With Russell               
Description :                                         
Organizer   :                                         
Location    :                                         
Cartegories :                                         
Priority    : 0
UID         : hlfhcpi0j360j8fteop49cvk68@google.com   
Start       : 2010-09-28 08:30:00 AM -0400            
End         : 2010-09-28 09:00:00 AM -0400            
Alarm       : 2010-09-28 04:20:00 AM -0400  
WARNING[5196]: res_calendar.c:648 do_notify: Channel should be in 
form Tech/Dest (was '')
channel=SIP/0000FFFF0001
app=Playback
appdata=this-is-yr-wakeup-call
*CLI> module reload res_calendar.so
    -- Dialing SIP/0000FFFF0001 for notification on calendar myGoogleCal
  == Using SIP RTP CoS mark 5
    -- Called 0000FFFF0001
    -- SIP/0000FFFF0001-00000001 is ringing
    -- SIP/0000FFFF0001-00000001 connected line has changed, passing it to 
       Calendar/myGoogleCal-5fd3c52
    -- SIP/0000FFFF0001-00000001 answered Calendar/myGoogleCal-5fd3c52
    -- <SIP/0000FFFF0001-00000001> Playing 'this-is-yr-wakeup-call.ulaw' 
       (language 'en')
[AutomatedMeetingSetup]
exten => start,1,Verbose(2,Triggering meeting setup for two participants)
   same => n,Set(DeviceToDial=${FILTER(0-9A-Za-z,${CALENDAR_EVENT(location)})})
   same => n,Dial(SIP/${DeviceToDial},30)
   same => n,Hangup()
This is where our calendar triggers a call to our device

    -- Dialing SIP/0000FFFF0001 for notification on calendar myGoogleCal
  == Using SIP RTP CoS mark 5
    -- Called 0000FFFF0001
    -- SIP/0000FFFF0001-00000004 is ringing

And now we have answered the call from Asterisk triggered by an event

    -- SIP/0000FFFF0001-00000004 connected line has changed, passing it to 
       Calendar/myGoogleCal-347ec99
    -- SIP/0000FFFF0001-00000004 answered Calendar/myGoogleCal-347ec99

Upon answer, we trigger some dialplan that looks up the endpoint to call

    -- Executing [start@AutomatedMeetingSetup:1] Verbose("SIP/0000FFFF0001-00000004", 
       "2,Triggering meeting setup for two participants") in new stack
  == Triggering meeting setup for two participants

This is where we used CALENDAR_EVENT(location) to get the remote device

    -- Executing [start@AutomatedMeetingSetup:2] Set("SIP/0000FFFF0001-00000004", 
       "DeviceToDial=0000FFFF0002") in new stack

And now we're dialing that endpoint

    -- Executing [start@AutomatedMeetingSetup:3] Dial("SIP/0000FFFF0001-00000004", 
       "SIP/0000FFFF0002,30") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 0000FFFF0002
    -- SIP/0000FFFF0002-00000005 is ringing

The other end answered the call, and Asterisk bridged us together

    -- SIP/0000FFFF0002-00000005 answered SIP/0000FFFF0001-00000004
    -- Locally bridging SIP/0000FFFF0001-00000004 and SIP/0000FFFF0002-00000005

[AutomatedMeetingSetup]
exten => start,1,Verbose(2,Triggering meeting setup for two participants)

; *** This line should not have any line breaks
   same => n,Read(CheckMeetingAcceptance,to-confirm-wakeup&press-1&otherwise
&press-2,,1)

   same => n,GotoIf($["${CheckMeetingAcceptance}" != "1"]?hangup,1)

   same => n,Playback(silence/1&pls-rcrd-name-at-tone&and-prs-pound-whn-finished)

; We set a random number and assign it to the end of the recording
; so that we have a unique filename in case this is used by multiple
; people at the same time.
;
; We also prefix it with a double underscore because the channel
; variable also needs to be available to the channel we're going to call
;
   same => n,Set(__RandomNumber=${RAND()})
   same => n,Record(/tmp/meeting-invite-${RandomNumber}.ulaw)

   same => n,Set(DeviceToDial=${FILTER(0-9A-Za-z,${CALENDAR_EVENT(location)})})
   same => n,Dial(SIP/${DeviceToDial},30,M(CheckConfirm))
   same => n,Hangup()

exten => hangup,1,Verbose(2,Call was rejected)
   same => n,Playback(vm-goodbye)
   same => n,Hangup()

[macro-CheckConfirm]
exten => s,1,Verbose(2,Allowing called party to accept or reject)
   same => n,Playback(/tmp/meeting-invite-${RandomNumber})

; *** This line should not have any line breaks
   same => n,Read(CheckMeetingAcceptance,to-confirm-wakeup&press-1&otherwise
&press-2,,1)

   same => n,GotoIf($["${CheckMeetingAcceptance}" != "1"]?hangup,1)

exten => hangup,1,Verbose(2,Call was rejected by called party)
   same => n,Playback(vm-goodbye)
   same => n,Hangup()
[AutomatedMeetingSetup]
exten => start,1,Verbose(2,Calling multiple people and placing into a conference)

; Get information from calendar and save that information. Prefix
; CalLocation with an underscore so it is available to the Local
; channel (variable inheritance).
;
   same => n,Set(CalDescription=${CALENDAR_EVENT(description)})
   same => n,Set(_CalLocation=${CALENDAR_EVENT(location)})
   same => n,Set(X=1)

; Our separator is a caret (^), so the description should be in the
; format of:    0000FFFF0001^0000FFFF0002^etc...
;
   same => n,Set(EndPoint=${CUT(CalDescription,^,${X})})

; This loop is used to build the ${ToDial} variable, which contains
; a list of Local channels to be dialed, thereby triggering the multiple
; Originate() actions simultaneously instead of linearly
;
   same => n,While($[${EXISTS(${EndPoint})}])

; This statement must be on a single line
   same => n,Set(ToDial=${IF($[${ISNULL(${ToDial})}]?
                     :${ToDial}&)}Local/${EndPoint}@MeetingOriginator)
   same => n,Set(X=$[${X} + 1])
   same => n,Set(EndPoint=${CUT(CalDescription,^,${X})})
   same => n,EndWhile()

; If no values are passed back, then don't bother dialing
   same => n,GotoIf($[${ISNULL(${ToDial})}]?hangup)
   same => n,Dial(${ToDial})

; After our Dial() statement returns, we should be placed into
; the conference room. We are marked, so the conference can start
; (which is indicated by the 'A' flag to MeetMe).
;
   same => n,MeetMe(${CalLocation},dA)
   same => n(hangup),Hangup()

[MeetingOriginator]
exten => _[A-Za-z0-9].,1,NoOp()
   same => n,Set(Peer=${FILTER(A-Za-z0-9,${EXTEN})})

; Originate calls to a peer as passed to us from the Local channel. Upon
; answer, the called party should execute the dialplan located at the
; _meetme-XXXX extension, where XXXX is the conference room number.
;
   same => n,Originate(SIP/${Peer},exten,MeetingOriginator,meetme-${CalLocation},1)
   same => n,Hangup()

; Join the meeting; using the 'w' flag, which means 'wait for marked
; user to join before starting'
;
exten => _meetme-XXXX,1,Verbose(2,Joining a meeting)
   same => n,Answer()
   same => n,MeetMe(${EXTEN:7},dw)
   same => n,Hangup()
exten => 3000,1,Verbose(2,Simple calendar busy check example)
   same => n,Set(CurrentExten=${EXTEN})
   same => n,Set(CalendarBusy=${CALENDAR_BUSY(myGoogleCal)})
   same => n,GotoIf($["${CalendarBusy}" = "1"]?voicemail,1)
   same => n,Dial(SIP/0000FFFF0002,30)
   same => n,Goto(voicemail,1)

exten => voicemail,1,Verbose(2,Caller sent to voicemail)

; *** This line should not have any line breaks
   same => n,GotoIf($["${DIALSTATUS}" = "BUSY" | 
"${CalendarBusy}" = "1"]?busy:unavail)

   same => n(busy),VoiceMail(${CurrentExten}@shifteight,b)
   same => n,Hangup()

   same => n(unavail),VoiceMail(${CurrentExten}@shifteight,u)
   same => n,Hangup()
exten => _3XXX,1,Verbose(2,Simple calendar busy check example)
   same => n,Set(CurrentExten=${EXTEN})
   same => n,GotoIf($[${DB_EXISTS(extension/${CurrentExten}/device)}]?:no_device,1)
   same => n,Set(CurrentDevice=${DB_RESULT})
   same => n,GotoIf($[${DB_EXISTS(extension/${CurrentExten}/calendar)}]?:no_calendar)
   same => n,Set(CalendarBusy=${CALENDAR_BUSY(${DB_RESULT})})
   same => n,GotoIf($[${CalendarBusy}]?voicemail,1)
   same => n(no_calendar),Verbose(2,No calendar was found for this user)
   same => n,Dial(SIP/${CurrentDevice},30)
   same => n,Goto(voicemail,1)

exten => voicemail,1,Verbose(2,Sending caller to voicemail)

; *** This line should not have any line breaks
   same => n,GotoIf($[${DB_EXISTS(extension/${CurrentExten}/voicemail_context)}]
?:no_voicemail)

   same => n,Set(VoiceMailContext=${DB_RESULT})
; *** This line should not have any line breaks
   same => n,Set(VoiceMailStatus=${IF($["${DIALSTATUS}" = "BUSY" | 
0${CalendarBusy}]?b:u)})
   same => n,VoiceMail(${CurrentExten}@${VoiceMailContext},${VoiceMailStatus})
   same => n,Hangup()

   same => n(no_voicemail),Playback(number-not-answering)
   same => n,Hangup()

exten => no_device,1,Verbose(2,No device found in the DB)
   same => n,Playback(invalid)
   same => n,Hangup()
(Calendar ID: 2hfb6p5974gds924j61cmg4gfd@group.calendar.google.com)
https://www.google.com/calendar/dav/<calendar_id>/events/
[phone_call_calendar]
type=caldav

; The URL must be on a single line
url=https://www.google.com/calendar/dav/
   shifteight.org_2hfb6p5974gds924j61cmg4gfd@group.calendar.google.com/events/

user = leif@shifteight.org
secret = my_secret_password
refresh=15
timeframe=120

*CLI> module reload res_calendar.so
*CLI> calendar show calendars
Calendar             Type       Status
--------             ----       ------
phone_call_calendar  caldav     free  
[LocalSets]
exten => _NXXNXXXXXX,1,Verbose(2,Outbound calls)
   same => n,Set(CalendarStart=${EPOCH}) ; Used by CALENDAR_WRITE()
   same => n,Set(X=${EXTEN})             ; Used by CALENDAR_WRITE()
   same => n,Dial(SIP/ITSP/${EXTEN},30)
   same => n,NoOp(Handle standard voicemail stuff here)
   same => n,Hangup()

exten => h,1,Verbose(2,Call cleanup)

; Everything that follows must be on a single line
   same => n,Set(CALENDAR_WRITE(phone_call_calendar,summary,description,start,end)=
      OUTBOUND: ${X},Phone call to ${X} lasted for ${CDR(billsec)} seconds.,
      ${CalendarStart},${EPOCH})
$ sudo yum install openssl-devel pam-devel
$ sudo apt-get install libssl-dev libpam0g-dev
$ cd ~/src/asterisk-complete
$ mkdir thirdparty
$ cd thirdparty
$ wget ftp://ftp.cac.washington.edu/mail/imap.tar.Z
$ tar zxvf imap.tar.Z
$ cd imap-2007e
$ make lnp EXTRACFLAGS="-fPIC -I/usr/include/openssl" IP6=4
$ make lnp EXTRACFLAGS="-I/usr/include/openssl" IP6=4
$ cd ~/src/asterisk-complete/asterisk/1.8.0
$ ./configure --with-imap=~/src/asterisk-complete/thirdparty/imap-2007e/
$ make menuselect
[general]
format=wav49              ; format to store files
imapserver=imap.gmail.com ; IMAP server location
imapport=993              ; port IMAP server listens to
imapflags=ssl             ; flags required for connecting
expungeonhangup=yes       ; delete messages on hangup
pollmailboxes=yes         ; used for message waiting indication
pollfreq=30               ; how often to check for message changes
[shifteight]
100 => 0107,Leif Madsen,leif@shifteight.org
101 => 0523,Jim VanMeggelen,jim@shifteight.org,,attach=no|maxmsg=100
102 => 11042,Tilghman Lesher,,,attach=no|tz=central
[shifteight]
100 => 0107,Leif Madsen,,,|imapuser=leif@shifteight.org|imappassword=secret
*CLI> core set debug 10
*CLI> module reload app_voicemail.so
DEBUG[3293]: app_voicemail.c:2734 mm_log: IMAP Info: Trying IP address [74.125.53.109]
DEBUG[3293]: app_voicemail.c:2734 mm_log: IMAP Info: Gimap ready for requests 
             from 99.228.XXX.XXX 13if2973206wfc.0
DEBUG[3293]: app_voicemail.c:2757 mm_login: Entering callback mm_login
DEBUG[3293]: app_voicemail.c:2650 mm_exists: Entering EXISTS callback for message 7
DEBUG[3293]: app_voicemail.c:3074 set_update: User leif@shifteight.org mailbox set for 
update.
DEBUG[3293]: app_voicemail.c:2510 init_mailstream: Before mail_open, server: 
             {imap.gmail.com:993/imap/ssl/user=leif@shifteight.org}INBOX, box:0
DEBUG[3293]: app_voicemail.c:2734 mm_log: IMAP Info: Reusing connection to 
             gmail-imap.l.google.com/user="leif@shifteight.org"

$ sudo yum install iksemel-devel openssl-devel
$ sudo apt-get install libiksemel-dev libssl-dev
[general]
debug=no
autoprune=no
autoregister=yes
auth_policy=accept

[asterisk]
type=client
serverhost=talk.google.com
username=asterisk@shifteight.org
secret=<super_secret_password>
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Ohai from Asterisk"
*CLI> jabber reload
Jabber Reloaded.
*CLI> jabber show connections
Jabber Users and their status:
       User: asterisk@shifteight.org    - Connected
----
   Number of users: 1
[LocalSets]
exten => 104,1,Answer()

; *** This line should not have any line breaks
   same => n,JabberSend(asterisk,jim@shifteight.org,Incoming call from 
${CALLERID(all)})

   same => n,Dial(SIP/0000FFFF0002,30)
   same => n,Hangup()
exten => 106,1,Answer()

   ; All text must be on a single line.
   same => n,JabberSend(asterisk,leif.madsen@gmail.com,Incoming call from
${CALLERID(all)}. Press 1 to route to desk. Press 2 to send to voicemail.)

   same => n,Set(JabberResponse=${JABBER_RECEIVE(asterisk,leif@shifteight.org)})
   same => n,GotoIf($["${JabberResponse}" = "1"]?dial,1)
   same => n,GotoIf($["${JabberResponse}" = "2"]?voicemail,1)
   same => n,Goto(dial,1)

exten => dial,1,Verbose(2,Calling our desk)
   same => n,Dial(SIP/0000FFFF0002,6)
   same => n,Goto(voicemail,1)

exten => voicemail,1,Verbose(2,VoiceMail)

; *** This line should not have any line breaks
   same => n,Set(VoiceMailStatus=${IF($[${ISNULL(${DIALSTATUS})} 
| "${DIALSTATUS}" = "BUSY"]?b:u)})

   same => n,Playback(silence/1)
   same => n,VoiceMail(100@lmentinc,${VoiceMailStatus})
   same => n,Hangup()

Set(JabberResponse=${JABBER_RECEIVE(asterisk,leif@shifteight.org,10)})
exten => 106,1,Verbose(2,Example using the Local channel)
   same => n,Dial(Local/jabber@${CONTEXT}/n&Local/dial@${CONTEXT}/n)

exten => jabber,1,Verbose(2,Send an XMPP message and expect a response)

; *** This line should not have any line breaks
   same => n,JabberSend(asterisk,leif.madsen@gmail.com,Incoming call from 
${CALLERID(all)}. Press 2 to send to voicemail.)

   same => n,Set(JabberResponse=${JABBER_RECEIVE(asterisk,leif@shifteight.org,6)})
   same => n,GotoIf($["${JabberResponse}" = "2"]?voicemail,1)
   same => n,Hangup()

exten => dial,1,Verbose(2,Calling our desk)
   same => n,Dial(SIP/0000FFFF0002,15)
   same => n,Goto(voicemail,1)

exten => voicemail,1,Verbose(2,VoiceMail)
   same => n,Answer()

; *** This line should not have any line breaks
   same => n,Set(VoiceMailStatus=${IF($[${ISNULL(${DIALSTATUS})} 
| "${DIALSTATUS}" = "BUSY"]?b:u)})

   same => n,Playback(silence/1)
   same => n,VoiceMail(100@lmentinc,${VoiceMailStatus})
   same => n,Hangup()

[general]
bindaddr=0.0.0.0        ; Address to bind to
allowguests=yes         ; Allow calls from people not in contact list

; Optional arguments
; externip=<external IP of server>
; stunaddr=<stun.yourdomain.tld>

[guest]                 ; special account for options on guest account
disallow=all                            
allow=ulaw
context=gtalk_incoming
connection=asterisk     ; connection name defined in jabber.conf
*CLI> module unload chan_gtalk.so
*CLI> module load chan_gtalk.so
[gtalk_incoming]
exten => s,1,Verbose(2,Incoming Gtalk call from ${CALLERID(all)})
   same => n,Answer()
   same => n,Dial(SIP/0000FFFF0001,30)
   same => n,Hangup()
[gtalk_incoming]
exten => my_asterisk_user,1,Verbose(2,Gtalk call from ${CALLERID(all)})
   same => n,Answer()
   same => n,Dial(SIP/0000FFFF0001,3)
   same => n,Hangup()
[gtalk_incoming]
exten => s,1,Verbose(2,Incoming call from ${CALLERID(all)})
   same => n,Answer()
   same => n,Wait(2)
   same => n,SendDTMF(2)
   same => n,Dial(SIP/0000FFFF0001,30)
   same => n,Hangup()
[LocalSets]
exten => 123,1,Verbose(2,Extension 123 calling some_user@gmail.com)
   same => n,Dial(Gtalk/asterisk/some_user@gmail.com,30)
   same => n,Hangup()
[LocalSets]
exten => _1NXXNXXXXXX,1,Verbose(2,Placing call to ${EXTEN} via Google Voice)
   same => n,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
   same => n,Hangup()
[general]
default_user=pbx.shifteight.org

[pbx.shifteight.org]
secret=my_secret_pass
context=skype_incoming
exten=start
buddy_autoadd=true
[LocalSets]
exten => 100,1,Answer()
   same => n,Dial(Skype/vuc.me,30)
   same => n,Playback(silence/1&user&is-curntly-unavail)
   same => n,Hangup()
[LocalSets]
exten => _SKYPE-.,1,Verbose(2,Dialing via Skype)
   same => n,Set(NameToDial=${FILTER(a-zA-Z0-9.,${EXTEN:6})})
   same => n,Playback(silence/1&pls-wait-connect-call)
   same => n,Dial(Skype/${NameToDial},30)
   same => n,Playback(user&is-curntly-unavail)
   same => n,Hangup()
[skype_incoming]
exten => start,1,Verbose(2,Incoming Skype Call)
   same => n,Answer()
   same => n,Dial(SIP/0000FFFF0001,30)
   same => n,Playback(user&is-curntly-unavail)
   same => n,Hangup()
[LocalSets]
exten => 104,1,Answer()

; *** This line should not have any line breaks
   same => n,SkypeChatSend(pbx.shifteight.org,tfot.madsen,Incoming call from 
${CALLERID(all)})

   same => n,Dial(SIP/0000FFFF0002,30)
   same => n,Hangup()
exten => 106,1,Answer()

   ; All text must be on a single line.
   same => n,SkypeChatSend(pbx.shifteight.org,tfot.madsen,Incoming call from
${CALLERID(all)}. Press 1 to route to desk. Press 2 to send to voicemail.)

   ; Wait for a response for 6 seconds.
   ; *** This line should not have any line breaks
   same => n,Set(SkypeResponse=
${SKYPE_CHAT_RECEIVE(pbx.shifteight.org,tfot.madsen,6)})

   same => n,GotoIf($["${SkypeResponse}" = "1"]?dial,1)
   same => n,GotoIf($["${SkypeResponse}" = "2"]?voicemail,1)
   same => n,Goto(dial,1)

exten => dial,1,Verbose(2,Calling our desk)
   same => n,Dial(SIP/0000FFFF0002,6)
   same => n,Goto(voicemail,1)

exten => voicemail,1,Verbose(2,VoiceMail)

; *** This line should not have any line breaks
   same => n,Set(VoiceMailStatus=${IF($[${ISNULL(${DIALSTATUS})} 
| "${DIALSTATUS}" = "BUSY"]?b:u)})

   same => n,Playback(silence/1)
   same => n,VoiceMail(100@lmentinc,${VoiceMailStatus})
   same => n,Hangup()

[LocalSets]
exten => 75973,1,Verbose(2,Read off list of Skype accounts)
   same => n,Answer()
   same => n,Set(ID=${SKYPE_BUDDIES(pbx.shifteight.org)})
   same => n(new_buddy),Set(ARRAY(buddy,status)=${SKYPE_BUDDY_FETCH(${ID})})
   same => n,GotoIf($[${ISNULL(${buddy})}]?no_more_buddies)
   same => n,Festival(${buddy} is ${status})
   same => n,Read(Answer,if-correct-press&digits/1&otherwise-press&digits/2,1)
   same => n,GotoIf($[${Answer} = 2]?new_buddy)
   same => n,Dial(Skype/${buddy},30)
   same => n,Playback(user&is-curntly-unavail)
   same => n,Hangup()

exten => no_more_buddies,1,Verbose(2,No more buddies to find)
   same => n,Playback(dir-nomore)
   same => n,Hangup()

$ cd ~/src/asterisk-complete/asterisk/1.8/contrib/scripts/
$ sudo cp asterisk.ldap-schema /etc/ldap/schema/asterisk.schema
$ sudo /etc/init.d/slapd restart
$ sudo ldapadd -Y EXTERNAL -H ldapi:/// -f asterisk.ldif
$ echo "my_secret_password" | md5sum
a7be810a28ca1fc0668effb4ea982e58  -
$ cat > astuser.ldif

dn: uid=rbryant,ou=people,dc=shifteight,dc=org
objectClass: inetOrgPerson
objectClass: posixAccount
objectClass: shadowAccount
objectClass: AsteriskSIPUser
uid: rbryant
sn: Bryant
givenName: Russell
cn: RussellBryant
displayName: Russell Bryant
uidNumber: 1001
gidNumber: 10001
userPassword: {md5}a7be810a28ca1fc0668effb4ea982e58
gecos: Russell Bryant
loginShell: /bin/bash
homeDirectory: /home/russell
shadowExpire: -1
shadowFlag: 0
shadowWarning: 7
shadowMin: 8
shadowMax: 999999
shadowLastChange: 10877
mail: russell.bryant@shifteight.org
postalCode: 31000
l: Huntsville
o: shifteight
title: Asterisk User
postalAddress:
initials: RB
AstAccountCallerID: Russell Bryant
AstAccountContext: LocalSets
AstAccountDTMFMode: rfc2833
AstAccountMailbox: 101@shifteight
AstAccountNAT: yes
AstAccountQualify: yes
AstAccountType: friend
AstAccountDisallowedCodec: all
AstAccountAllowedCodec: ulaw
AstAccountMusicOnHold: default

Ctrl+D
$ sudo ldapadd -x -D cn=admin,dc=shifteight,dc=org -f astusers.ldif -W
Enter LDAP Password: 
adding new entry "uid=rbryant,ou=people,dc=shifteight,dc=org"
$ sudo apt-get install openldap-dev
$ sudo yum install openldap-devel
[_general]
url=ldap://172.16.0.103:389
protocol=3
basedn=dc=shifteight,dc=org
user=cn=admin,dc=shifteight,dc=org
pass=canada

[sip]
name = cn
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
...
lastms = AstAccountLastQualifyMilliseconds
useragent = AstAccountUserAgent
additionalFilter=(objectClass=AsteriskSIPUser)
          ldap,"ou=people,dc=shifteight,dc=org",sip

sipusers => ldap,"ou=people,dc=shifteight,dc=org",sip
sippeers => ldap,"ou=people,dc=shifteight,dc=org",sip
rtcachefriends=yes
$ sudo yum install festival
$ sudo apt-get install festival
(define (tts_textasterisk string mode)
"(tts_textasterisk STRING MODE)
Apply tts to STRING. This function is specifically designed for
use in server mode so a single function call may synthesize the string.
This function name may be added to the server safe functions."
(let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string)))))
(utt.wave.resample wholeutt 8000)
(utt.wave.rescale wholeutt 5)
(utt.send.wave.client wholeutt)))
$ sudo festival_server 2>&1 > /dev/null &
$ cp ~/asterisk-complete/asterisk/1.8/configs/festival.conf.sample \
/etc/asterisk/festival.conf
*CLI> core show application festival
*CLI> module load app_festival.so
[LocalSets]
exten => 203,1,Verbose(2,This is a Festival test)
   same => n,Answer()
   same => n,Playback(silence/1)
   same => n,Festival(Hello World)
   same => n,Hangup()
exten => 202,1,Verbose(2,Trying out Festival)
   same => n,Answer()

; *** This line should not have any line breaks
   same => n,System(echo "This is a test of Festival" 
| /usr/bin/text2wave -scale 1.5 -F 8000 -o /tmp/festival.wav)

   same => n,Playback(/tmp/festival)
   same => n,System(rm -f /tmp/festival.wav)
   same => n,Hangup()
$ mkdir ~/src/asterisk-complete/thirdparty
$ cd ~/src/asterisk-complete/thirdparty
$ wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre17.tgz
$ tar zxvf spandsp-0.0.6pre17.tgz
$ cd spandsp-0.0.6
$ ./configure
$ make
$ sudo make install
$ sudo cat >> /etc/ld.so.conf.d/usrlocallib.conf
/usr/local/lib

$ sudo ldconfig
$ cd ~/src/asterisk-complete/asterisk/1.8/
$ ./configure
$ make menuselect
[*] res_fax_spandsp
XXX res_fax_spandsp
$ make
$ make install
*CLI> module show like res_fax_spandsp.so
noload => res_fax_spandsp.so

noload => res_fax_digium.so
exten => fax,1,Verbose(3,Incoming fax)
; folder where your incoming faxes will initially be stored
  same => n,Set(FAXDEST=/tmp) 

; put a timestamp on this call so the resulting file is unique
  same => n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)}) 
  same => n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
  same => n,Verbose(3,- Fax receipt completed with status: ${FAXSTATUS})

; *** This line should not have any line breaks
  same => n,System(echo | mutt -a ${FAXDEST}/${tempfax} 
-s "received fax" somebody@shifteight.org)

exten => faxthis,1,SendFAX(/path/to/fax/file,d)
exten => faxthis,1,Verbose(2,Set options and transmit fax)

; some folder where your outgoing faxes will be found
  same => n,Set(faxlocation=/tmp) 

; In production you would probably not want to hardcode the filename
  same => n,Set(faxfile=faxfile.tif) 
  same => n,Set(FAXOPT(headerinfo)=Fax from ShiftEight.org)
  same => n,Set(FAXOPT(localstationid=4169671111)
  same => n,SendFax(${faxlocation}/${faxfile})

$ sudo yum -y install ghostscript
$ sudo apt-get install ghostscript
$ gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter -sOutputFile=<dest> <src>
fax:   "| /path/to/program/that/will/handle/incoming/fax/emails"
fax:   "| /asteriskpbx/fax.py"
#!/usr/bin/env python
"""Poor Man's Email to Fax Gateway.

This is a proof of concept email to fax gateway.  There are multiple aspects
that would have to be improved for it to be used in a production environment.

Copyright (C) 2010 - Russell Bryant, Leif Madsen, Jim Van Meggelen
Asterisk: The Definitive Guide
"""

import sys
import os
import email
import base64
import shutil
import socket

AMI_HOST = "localhost"
AMI_PORT = 5038
AMI_USER = "hello"
AMI_PASS = "world"

# This script will pull a TIFF out of an email and save it off to disk to allow
# the SendFax() application in Asterisk to send it.  This is the location on
# disk where the TIFF will be stored.
TIFF_LOCATION = "/tmp/loremipsum.tif"

# Read an email from stdin and parse it.
msg = email.message_from_file(sys.stdin)

# For testing purposes, if you wanted to read an email from a file, you could
# do this, instead.
#try:
#    f = open("email.txt", "r")
#    msg = email.message_from_file(f)
#    f.close()
#except IOError:
#    print "Failed to open email input file."
#    sys.exit(1)

# This next part pulls out a TIFF file attachment from the email and saves it
# off to disk in a format that can be used by the SendFax() application.  This
# part of the script is incredibly non-flexible.  It assumes that the TIFF file
# will be in a specific location in the structure of the message (the second
# part of the payload, after the main body).  Further, it assumes that the
# encoding of the TIFF attachment is base64.  This was the case for the test
# email that we were using that we generated with mutt.  Emails sent by users'
# desktop email clients will vary in _many_ ways.  To be used with user-
# generated emails, this section would have to be much more flexible.
try:
    f2 = open(TIFF_LOCATION, "w")
    f2.write(base64.b64decode(msg.get_payload()[1].get_payload().replace("\n", "")))
    f2.close()
except IOError:
    print "Failed to open file for saving off TIFF attachment."
    sys.exit(1)

# Now that we have a TIFF file to fax, connect to the Asterisk Manager Interface
# to originate a call.
ami_commands = """Action: Login\r
Username: %s\r
Secret: %s\r
\r
Action: Originate\r
Channel: Local/s@sendfax/n\r
Context: receivefax\r
Extension: s\r
Priority: 1\r
SetVar: SUBJECT=%s\r
\r
Action: Logoff\r
\r
""" % (AMI_USER, AMI_PASS, msg['subject'])

print ami_commands

def my_send(s, data):
    """Ensure that we send out the whole data buffer.
    """
    sent = 0
    while sent < len(data):
        res = s.send(data[sent:])
        if res == 0:
            break
        sent = sent + res

def my_recv(s):
    """Read input until there is nothing else to read.
    """
    while True:
        res = s.recv(4096)
        if len(res) == 0:
            break
        print res

s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.connect((AMI_HOST, AMI_PORT))
my_send(s, ami_commands)
my_recv(s)
s.shutdown(socket.SHUT_RDWR)
s.close()

faxbuffers => 12,half
faxdetect = both
;
; Turn on the AMI and ask it to only accept connections from localhost.
;
[general]
enabled = yes
webenabled = yes
bindaddr = 127.0.0.1

;
; Create an account called "hello", with a password of "world"
;
[hello]
secret=world

;
; Enable the built-in HTTP server, and only listen for connections on localhost.
;
[general]
enabled = yes
bindaddr = 127.0.0.1
$ telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.1
Action: Login
Username: hello
Secret: world

Response: Success
Message: Authentication accepted

Action: Ping

Response: Success
Ping: Pong
Timestamp: 1282739190.454046

Action: Logoff

Response: Goodbye
Message: Thanks for all the fish.

Connection closed by foreign host.
$ wget "http://localhost:8088/rawman?action=login&username=hello&secret=world" \
> --save-cookies cookies.txt -O -

--2010-08-31 12:34:23--
Resolving localhost... 127.0.0.1
Connecting to localhost|127.0.0.1|:8088... connected.
HTTP request sent, awaiting response... 200 OK
Length: 55 [text/plain]
Saving to: `STDOUT'

Response: Success
Message: Authentication accepted

2010-08-31 12:34:23 (662 KB/s) - written to stdout [55/55]

$ wget "http://localhost:8088/rawman?action=ping" --load-cookies cookies.txt -O -

--2010-08-31 12:34:23--
Resolving localhost... 127.0.0.1
Connecting to localhost|127.0.0.1|:8088... connected.
HTTP request sent, awaiting response... 200 OK
Length: 63 [text/plain]
Saving to: `STDOUT'

Response: Success
Ping: Pong
Timestamp: 1283258063.040293

2010-08-31 12:34:23 (775 KB/s) - written to stdout [63/63]

$ wget "http://localhost:8088/rawman?action=logoff" --load-cookies cookies.txt -O -

--2010-08-31 12:34:23--
Resolving localhost... 127.0.0.1
Connecting to localhost|127.0.0.1|:8088... connected.
HTTP request sent, awaiting response... 200 OK
Length: 56 [text/plain]
Saving to: `STDOUT'

Response: Goodbye
Message: Thanks for all the fish.

2010-08-31 12:34:23 (696 KB/s) - written to stdout [56/56]

Header1: This is the first header<CR><LF>
Header2: This is the second header<CR><LF>
Header3: This is the last header of this message<CR><LF>
<CR><LF>
          Event: Hangup
Privilege: call,all
Channel: SIP/0004F2060EB4-00000000
Uniqueid: 1283174108.0
CallerIDNum: 2565551212
CallerIDName: Russell Bryant
Cause: 16
Cause-txt: Normal Clearing

Action: Login
Username: russell
Secret: russell

Response: Success
Message: Authentication accepted
$ curl -v "http://localhost:8088/rawman?action=login&username=hello&secret=world"

* About to connect() to localhost port 8088 (#0)
*   Trying 127.0.0.1... connected
* Connected to localhost (127.0.0.1) port 8088 (#0)
> GET /rawman?action=login&username=hello&secret=worlda HTTP/1.1
> User-Agent: curl/7.19.7 (x86_64-pc-linux-gnu) libcurl/7.19.7 
OpenSSL/0.9.8k zlib/1.2.3.3 libidn/1.15
> Host: localhost:8088
> Accept: */*
> 
< HTTP/1.1 200 OK
< Server: Asterisk/1.8.0-beta4
< Date: Tue, 07 Sep 2010 11:51:28 GMT
< Connection: close
< Cache-Control: no-cache, no-store
< Content-Length: 55
< Content-type: text/plain
< Cache-Control: no-cache;
< Set-Cookie: mansession_id="0e929e60"; Version=1; Max-Age=60
< Pragma: SuppressEvents
< 

Response: Success
Message: Authentication accepted
* Closing connection #0

$ curl -v "http://localhost:8088/manager?action=login&username=hello&secret=world"

* About to connect() to localhost port 8088 (#0)
*   Trying 127.0.0.1... connected
* Connected to localhost (127.0.0.1) port 8088 (#0)
> GET /manager?action=login&username=hello&secret=world HTTP/1.1
> User-Agent: curl/7.19.7 (x86_64-pc-linux-gnu) libcurl/7.19.7 
OpenSSL/0.9.8k zlib/1.2.3.3 libidn/1.15
> Host: localhost:8088
> Accept: */*
> 
< HTTP/1.1 200 OK
< Server: Asterisk/1.8.0-beta4
< Date: Tue, 07 Sep 2010 12:19:05 GMT
< Connection: close
< Cache-Control: no-cache, no-store
< Content-Length: 881
< Content-type: text/html
< Cache-Control: no-cache;
< Set-Cookie: mansession_id="139deda7"; Version=1; Max-Age=60
< Pragma: SuppressEvents
< 

<title>Asterisk&trade; Manager Interface</title><body bgcolor="#ffffff">
<table align=center bgcolor="#f1f1f1" width="500">
<tr><td colspan="2" bgcolor="#f1f1ff"><h1>Manager Tester</h1></td></tr>
<tr><td colspan="2" bgcolor="#f1f1ff"><form action="manager" method="post">
	Action: <select name="action">
		<option value="">-----&gt;</option>
		<option value="login">login</option>
		<option value="command">Command</option>
		<option value="waitevent">waitevent</option>
		<option value="listcommands">listcommands</option>
	</select>
	or <input name="action"><br/>
	CLI Command <input name="command"><br>
	user <input name="username"> pass <input type="password" name="secret"><br>
	<input type="submit">
</form>
</td></tr>
<tr><td>Response</td><td>Success</td></tr>
<tr><td>Message</td><td>Authentication accepted</td></tr>
<tr><td colspan="2"><hr></td></tr>
* Closing connection #0
</table></body
$ curl -v "http://localhost:8088/mxml?action=login&username=hello&secret=world"

* About to connect() to localhost port 8088 (#0)
*   Trying 127.0.0.1... connected
* Connected to localhost (127.0.0.1) port 8088 (#0)
> GET /mxml?action=login&username=hello&secret=world HTTP/1.1
> User-Agent: curl/7.19.7 (x86_64-pc-linux-gnu) libcurl/7.19.7 
OpenSSL/0.9.8k zlib/1.2.3.3 libidn/1.15
> Host: localhost:8088
> Accept: */*
> 
< HTTP/1.1 200 OK
< Server: Asterisk/1.8.0-beta4
< Date: Tue, 07 Sep 2010 12:26:58 GMT
< Connection: close
< Cache-Control: no-cache, no-store
< Content-Length: 146
< Content-type: text/xml
< Cache-Control: no-cache;
< Set-Cookie: mansession_id="536d17a4"; Version=1; Max-Age=60
< Pragma: SuppressEvents
< 

<ajax-response>
<response type='object' id='unknown'>
<generic response='Success' message='Authentication accepted' />
</response>
* Closing connection #0
</ajax-response>
$ wget --save-cookies cookies.txt \
> "http://localhost:8088/mxml?action=login&username=hello&secret=world" -O -

<ajax-response>
<response type='object' id='unknown'>
    <generic response='Success' message='Authentication accepted' />
</response>
</ajax-response>

$ wget --load-cookies cookies.txt "http://localhost:8088/mxml?action=waitevent" -O -

<ajax-response>
<response type='object' id='unknown'>
    <generic response='Success' message='Waiting for Event completed.' />
</response>
<response type='object' id='unknown'>
    <generic event='PeerStatus' privilege='system,all' 
             channeltype='SIP' peer='SIP/0000FFFF0004' 
             peerstatus='Registered' address='172.16.0.160:5060' />
</response>
<response type='object' id='unknown'>
    <generic event='WaitEventComplete' />
</response>
</ajax-response>

$ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0
exten => 500,1,AGI(hello-world.sh)
#!/bin/bash

# Consume all variables sent by Asterisk
while read VAR && [ -n ${VAR} ] ; do : ; done

# Answer the call.
echo "ANSWER"
read RESPONSE

# Say the letters of "Hello World"
echo 'SAY ALPHA "Hello World" ""'
read RESPONSE

exit 0

*CLI> agi set debug on
AGI Debugging Enabled

    -- Executing [500@phones:1] AGI("SIP/0004F2060EB4-00000009", 
       "hello-world.sh") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hello-world.sh
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_request: hello-world.sh
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_channel: SIP/0004F2060EB4-00000009
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_language: en
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_type: SIP
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_uniqueid: 1284382003.9
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_version: 1.8.0-beta4
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_callerid: 2563619899
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_calleridname: Russell Bryant
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_callingpres: 0
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_callingani2: 0
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_callington: 0
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_callingtns: 0
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_dnid: 7010
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_rdnis: unknown
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_context: phones
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_extension: 500
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_priority: 1
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_enhanced: 0.0
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_accountcode: 
<SIP/0004F2060EB4-00000009>AGI Tx >> agi_threadid: 140071216785168
<SIP/0004F2060EB4-00000009>AGI Tx >> 
<SIP/0004F2060EB4-00000009>AGI Rx << ANSWER
<SIP/0004F2060EB4-00000009>AGI Tx >> 200 result=0
<SIP/0004F2060EB4-00000009>AGI Rx << SAY ALPHA "Hello World" ""
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/h.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/e.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/l.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/l.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/o.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/space.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/w.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/o.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/r.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/l.gsm' (language 'en')
    -- <SIP/0004F2060EB4-00000009> Playing 'letters/d.gsm' (language 'en')
<SIP/0004F2060EB4-00000009>AGI Tx >> 200 result=0
    -- <SIP/0004F2060EB4-00000009>AGI Script hello-world.sh completed, returning 0

AGI(command[,arg1[,arg2[,...]]])
*CLI> core show application AGI
*CLI> core show application EAGI
exten => 1234,1,AGI(agi://127.0.0.1)
exten => 1234,1,AGI(agi://127.0.0.1:4574)
exten => 1234,1,AGI(agi://192.168.1.199,arg1,arg2,arg3)
exten => 1234,1,AGI(hagi://shifteight.org)
exten => 1234,1,AGI(agi:async)
;
; Define a user called 'hello', with a password of 'world'.
; Give this user read/write permissions for AGI.
;
[hello]
secret = world
read = agi
write = agi
agi_variable: value
Event: AsyncAGI
Privilege: agi,all
SubEvent: Start
Channel: SIP/0000FFFF0001-00000000
Env: agi_request%3A%20async%0Aagi_channel%3A%20SIP%2F0000FFFF0001-00000000%0A \
     agi_language%3A%20en%0Aagi_type%3A%20SIP%0Aagi_uniqueid%3A%201285219743.0%0A \
     agi_version%3A%201.8.0-beta5%0Aagi_callerid%3A%2012565551111%0A \
     agi_calleridname%3A%20Julie%20Bryant%0Aagi_callingpres%3A%200%0A \
     agi_callingani2%3A%200%0Aagi_callington%3A%200%0Aagi_callingtns%3A%200%0A \
     agi_dnid%3A%20111%0Aagi_rdnis%3A%20unknown%0Aagi_context%3A%20LocalSets%0A \
     agi_extension%3A%20111%0Aagi_priority%3A%201%0Aagi_enhanced%3A%200.0%0A \
     agi_accountcode%3A%20%0Aagi_threadid%3A%20-1339524208%0A%0A

200 result=0
exten => 7011,1,AGI(agi:async)

Action: AGI
Channel: SIP/0004F2060EB4-00000013
ActionID: my-action-id
CommandID: my-command-id
Command:  VERBOSE "Puppies like cotton candy." 1

Response: Success
ActionID: my-action-id
Message: Added AGI command to queue

Event: AsyncAGI
Privilege: agi,all
SubEvent: Exec
Channel: SIP/0004F2060EB4-00000013
CommandID: my-command-id
Result: 200%20result%3D1%0A

    -- Executing [7011@phones:1] AGI("SIP/0004F2060EB4-00000013", 
       "agi:async") in new stack
 agi:async: Puppies like cotton candy.
  == Spawn extension (phones, 7011, 1) exited non-zero on 'SIP/0004F2060EB4-00000013'

;
; Don't send SIGHUP to an AGI process
; or the "HANGUP" string to a FastAGI server.
;
exten => 500,1,Set(AGISIGHUP=no)
    same => n,AGI(my-agi-application)
Action: AGI
Channel: SIP/0004F2060EB4-0000001b
ActionID: my-action-id
CommandID: my-command-id
Command: ASYNCAGI BREAK

Response: Success
ActionID: my-action-id
Message: Added AGI command to queue

Event: AsyncAGI
Privilege: agi,all
SubEvent: End
Channel: SIP/0004F2060EB4-0000001b

; DUNDi configuration file for Toronto
;
[general]
;
department=IT
organization=toronto.example.com
locality=Toronto
stateprov=ON
country=CA
email=support@toronto.example.com
phone=+14165551212
;
; Specify bind address and port number.  Default is port 4520.
;bindaddr=0.0.0.0
port=4520
entityid=FF:FF:FF:FF:FF:FF
ttl=32
autokill=yes
;secretpath=dundi
; DUNDi configuration file for Vancouver
;
[general]
;
department=IT
organization=vancouver.example.com
locality=Vancouver
stateprov=BC
country=CA
email=support@vancouver.example.com
phone=+16135551212
;
; Specify bind address and port number.  Default port is 4520.
;bindaddr=0.0.0.0
port=4520
entityid=00:00:00:00:00:00
ttl=32
autokill=yes
;secretpath=dundi
[00:00:00:00:00:00] ; Vancouver Remote Office
model = symmetric
host = vancouver.example.com
inkey = vancouver
outkey = toronto
qualify = yes
dynamic=yes
$ cd /var/lib/asterisk/keys
$ sh ~/src/asterisk-complete/asterisk/1.8/contrib/scripts/astgenkey -n toronto
[FF:FF:FF:FF:FF:FF] ; Toronto Remote Office
model = symmetric
host = toronto.example.com
inkey = toronto
outkey = vancouver
qualify = yes
dynamic=yes
toronto*CLI> module reload res_crypto.so 
    -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
    -- Loaded PUBLIC key 'vancouver'
    -- Loaded PUBLIC key 'toronto'
    -- Loaded PRIVATE key 'toronto'
vancouver*CLI> module reload res_crypto.so 
    -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
    -- Loaded PUBLIC key 'toronto'
    -- Loaded PUBLIC key 'vancouver'
    -- Loaded PRIVATE key 'vancouver'
*CLI> keys show
Key Name           Type     Status           Sum                              
------------------ -------- ---------------- -------------------------------- 
vancouver          PRIVATE  [Loaded]         c02efb448c37f5386a546f03479f7d5e 
vancouver          PUBLIC   [Loaded]         0a5e53420ede5c88de95e5d908274fb1 
toronto            PUBLIC   [Loaded]         5f806860e0c8219f597f876caa6f2aff 

3 known RSA keys.
*CLI> module reload pbx_dundi.so
    -- Reloading module 'pbx_dundi.so' (Distributed Universal Number 
       Discovery (DUNDi))
  == Parsing '/etc/asterisk/dundi.conf': Found
toronto*CLI> dundi show peers
EID                  Host                Port   Model      AvgTime  Status         
00:00:00:00:00:00    172.16.0.104    (S) 4520   Symmetric  Unavail  OK (3 ms)      
1 dundi peers [1 online, 0 offline, 0 unmonitored]

dundi_context => local_context,weight,technology,destination[,options]]
[mappings]

; All on a single line
;
extensions => RegisteredDevices,0,SIP,dundi:very_secret_secret@toronto.example.com/
${NUMBER},nopartial
[mappings]

; All on a single line
;
extensions => RegisteredDevices,0,SIP,dundi:very_secret_secret@vancouver.example.com/
${NUMBER},nopartial
[RegisteredDevices]
exten => 1000,1,NoOp()
*CLI> dialplan reload

*CLI> module reload pbx_dundi.so
    -- Reloading module 'pbx_dundi.so' (Distributed Universal Number 
       Discovery (DUNDi))
  == Parsing '/etc/asterisk/dundi.conf':   == Found
toronto*CLI> dundi show mappings
DUNDi Cntxt  Weight  Local Cntxt  Options    Tech  Destination              
extensions   0       RegisteredDe NONE       SIP   dundi:${SECRET}@172.16.0.

[00:00:00:00:00:00] ; Vancouver Remote Office
model = symmetric
host = vancouver.example.com
inkey = vancouver
outkey = toronto
qualify = yes
dynamic=yes
permit=extensions
include=extensions
[FF:FF:FF:FF:FF:FF] ; Toronto Remote Office
model = symmetric
host = toronto.example.com
inkey = toronto
outkey = vancouver
qualify = yes
dynamic=yes
permit=extensions
include=extensions
*CLI> module reload pbx_dundi.so
*CLI> dundi show peer 00:00:00:00:00:00
Peer:    00:00:00:00:00:00
Model:   Symmetric
Host:    172.16.0.104
Port:    4520
Dynamic: no
Reg:     No
In Key:  vancouver
Out Key: toronto
Include logic:
-- include extensions
Query logic:
-- permit extensions
vancouver*CLI> dundi lookup 1000@extensions bypass
  1.     0 SIP/dundi:very_secret_secret@172.16.0.161/1000 (EXISTS)
     from ff:ff:ff:ff:ff:ff, expires in 3600 s
DUNDi lookup completed in 12 ms
[dundi]
type=user
secret=very_secret_secret
context=DUNDi_Incoming
disallow=all
allow=ulaw
allow=alaw
toronto*CLI> sip reload
[DUNDi_Incoming]
exten => 1000,1,Verbose(2,Incoming call from the DUNDi peer)
   same => n,Answer()
   same => n,Playback(silence/1)
   same => n,Playback(tt-weasels)
   same => n,Hangup()
[LocalSets]
exten => 1000,1,Verbose(2,Test extension to place call to remote server)
same => n,Dial(SIP/dundi:very_secret_secret@172.16.0.161/1000,30)
same => n,Hangup()
[dundi]
type=friend
context=DUNDi_Incoming
dbsecret=dundi/secret
disallow=all
allow=ulaw
allow=alaw
[mappings]

; All on a single line
;
extensions => RegisteredDevices,0,SIP,dundi:${SECRET}@vancouver.example.com/
${NUMBER},nopartial
[RegisteredDevices]
exten => 1000,1,NoOp()

exten => 1001,1,NoOp()

exten => 1002,1,NoOp()
[RegisteredDevices]
exten => _416NXXXXXX,1,NoOp()
exten => _647NXXXXXX,1,NoOp()
exten => _905NXXXXXX,1,NoOp()
[RegisteredDevices]
exten => _1[1-3]XX,1,NoOp()  ; extensions 1100->1399

exten => _1[7-9]XX,1,NoOp()  ; extensions 1700->1999
[general]
regcontext=RegisteredDevices

[0000FFFF0001](office-phone)
regexten=1001
*CLI> dialplan show RegisteredDevices
[ Context 'RegisteredDevices' created by 'SIP' ]
  '1001' =>         1. Noop(0000FFFF0001)                   [SIP]
  '1002' =>         1. Noop(0000FFFF0002)                   [SIP]
*CLI> database put phones 1001/device 0000FFFF0001
Updated database successfully
[mappings]
; The mapping exists on a single line
extensions => RegisteredDevices,0,SIP,
   dundi:very_secret_secret@toronto.example.com/${NUMBER},nopartial
${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)}
vancouver*CLI> dundi lookup 1001@extensions bypass
  1.     0 SIP/dundi:very_awesome_password/0000FFFF0001 (EXISTS)
     from ff:ff:ff:ff:ff:ff, expires in 3600 s
DUNDi lookup completed in 77 ms
[TestContext]
exten => 1001,1,Verbose(2,Look up extension 1001)
same => n,Set(DUNDi_Result=${DUNDILOOKUP(1001,extensions,b)})
same => n,Verbose(2,The result of the lookup was ${DUNDi_Result})
same => n,Hangup()
[TestContext]
exten => _1XXX,1,Verbose(2,Looking up results for extension ${EXTEN})

; Perform our lookup and save the resulting ID to DUNDI_ID
   same => n,Set(DUNDI_ID=${DUNDIQUERY(${EXTEN},extensions,b)})
   same => n,Verbose(2,Showing all results returned from the DUNDi Query)

; The DUNDIRESULT() function can return the number of results using 'getnum'
   same => n,Set(NumberOfResults=${DUNDIRESULT(${DUNDI_ID},getnum)})
   same => n,Set(ResultCounter=1)

; If there is less than 1 result, no results were returned
   same => n,GotoIf($[${NumberOfResults} < 1]?NoResults,1)

; The start of our loop showing the returned values
   same => n,While($[${ResultCounter} <= ${NumberOfResults}])

; Save the returned result at position ${ResultCounter} to thisResult
   same => n,Set(thisResult=${DUNDIRESULT(${DUNDI_ID},${ResultCounter})})

; Show the current result on the console
   same => n,Verbose(2,One of the results returned was: ${thisResult})

; Increase the counter by one
   same => n,Set(ResultCounter=${INC(ResultCounter)})

; End of our loop
   same => n,EndWhile()
   same => n,Playback(silence/1)
   same => n,Playback(vm-goodbye)
   same => n,Hangup()

; If no results were found, execute this dialplan
exten => NoResults,1,Verbose(2,No results were found)
   same => n,Playback(silence/1)
   same => n,Playback(invalid)
   same => n,Hangup()
[subLookupExtension]
exten => _1XXX,1,Verbose(2,Looking up results for extension ${EXTEN})

; Perform our lookup and save the resulting ID to DUNDI_ID
   same => n,Set(DUNDI_ID=${DUNDIQUERY(${EXTEN},extensions,b)})
   same => n,Set(NumberOfResults=${DUNDIRESULT(${DUNDI_ID},getnum)})
   same => n,Set(ResultCounter=1)

; If no results are found, return 'None'
   same => n,GotoIf($[${NumberOfResults} < 1]?NoResults,1)

; Perform our loop
   same => n,While($[${ResultCounter} <= ${NumberOfResults}])

; Get the current value
   same => n,Set(thisResult=${DUNDIRESULT(${DUNDI_ID},${ResultCounter})})

; If the current value returned is not None, we have a resulting
; location to call and we can exit the loop
   same => n,ExecIf($["${thisResult}" != "None"]?ExitWhile())

; If we made it this far, no value has been returned yet that we want to
; use, so increase the counter and try the next value.
   same => n,Set(ResultCounter=${INC(ResultCounter)})

; End of our loop
   same => n,EndWhile()

; We've made it here because we made it to the end of the loop or we found
; a value we want to return. Check to see which it is. If we just ran out of
; values, return 'None'.
;
   same => n,GotoIf($["${thisResult}" = "None"]?NoResults,1)

; If we make it here, we have a value we want to return.
   same => n,Return(${thisResult})

; If there were no acceptable results, return the value 'None'
   exten => NoResults,1,Verbose(2,No results were found)
same => n,Return(None)
filename => type[,type[,type[,...]]]
[general]

[logfiles]
console => notice,warning,error,dtmf
messages => notice,warning,error
;verbose => notice,warning,error,verbose
$ asterisk -rx 'logger reload'
*CLI> logger reload
verbose => notice,warning,error,verbose
$ tail -1000 verbose

[Mar 11 09:38:35] VERBOSE[31362] logger.c:     -- IAX2/shifteight-4 answered Zap/1-1
[Mar 11 09:39:35] VERBOSE[2973] logger.c:     -- Starting simple switch on 'Zap/1-1'
[Mar 11 09:39:35] VERBOSE[31362] logger.c:   == Spawn extension (shifteight, s, 1) 
exited non-zero on 'Zap/1-1'
[Mar 11 09:39:35] VERBOSE[2973] logger.c:     -- Hungup 'Zap/1-1'
[Mar 11 09:39:35] VERBOSE[3680] logger.c:     -- Starting simple switch on 'Zap/1-1'
[Mar 11 09:39:35] VERBOSE[31362] logger.c:     -- Hungup 'Zap/1-1'
$ grep 31362 verbose
[Mar 11 09:38:35] VERBOSE[31362] logger.c:     -- IAX2/shifteight-4 answered Zap/1-1
[Mar 11 09:39:35] VERBOSE[31362] logger.c:   == Spawn extension (shifteight, s, 1) 
exited non-zero on 'Zap/1-1'
[Mar 11 09:39:35] VERBOSE[31362] logger.c:     -- Hungup 'Zap/1-1'

syslog.local0 => notice,warning,error ; or whatever type(s) you want to log
local0.*      /var/log/asterisk/syslog
*CLI> logger show channels
Channel                     Type     Status  Configuration
-------                     ----     ------  -------------
syslog.local0               Syslog   Enabled  - NOTICE WARNING ERROR VERBOSE
/var/log/asterisk/verbose   File     Enabled  - NOTICE WARNING ERROR VERBOSE
/var/log/asterisk/messages  File     Enabled  - NOTICE WARNING ERROR
                            Console  Enabled  - NOTICE WARNING ERROR DTMF

exten => 115,1,Verbose(Call start time: ${CDR(start)})
   same => n,Set(CDR(userfield)=zombie pancakes)
exten => 115,1,NoOp()
   same => n,Set(CDR(mycustomfield)=coffee)
   same => n,Verbose(I need some more ${CDR(mycustomfield)})

*CLI> core show function CDR
*CLI> core show applications like CDR
    -= Matching Asterisk Applications =-
               ForkCDR: Forks the Call Data Record. 
                 NoCDR: Tell Asterisk to not maintain a CDR for the current call 
              ResetCDR: Resets the Call Data Record. 
    -= 3 Applications Matching =-

*CLI> core show application <application name>
exten => 105,n,Set(CDR(useragent)=${CHANNEL(useragent)})
[mytable]

connection = asterisk
table = asterisk_cdr
alias <CDR variable> => <column name>
alias src => source
filter <CDR variable> => <content>
filter accountcode => 123
static <"Static Content Goes Here"> => <column name>
static "My Content" => my_identifier
<accountcode>,<src>,<dst>,<dcontext>,<clid>,<channel>,<dstchannel>,<lastapp>, \
    <lastadata>,<start>,<answer>,<end>,<duration>,<billsec>,<disposition>, \
    <amaflags>[,<uniqueid>][,<userfield>]

[mappings]

Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
   ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
   ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
   ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
   ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
   ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
   ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
   ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
   ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}
[general]

enabled = yes
<CDR variable> => <Header name>
[mappings]

rate => Rate
carrier => Carrier
exten => 110,1,Answer()
    same => n,Set(CDR(rate)=0.02)
    same => n,Set(CDR(carrier)=BS&S)
    same => n,Hangup()

*CLI> console dial 110@testing
Event: Cdr
Privilege: cdr,all
AccountCode: 
Source: 
Destination: 110
DestinationContext: testing
CallerID: 
Channel: Console/dsp
DestinationChannel: 
LastApplication: Hangup
LastData: 
StartTime: 2010-08-23 08:27:21
AnswerTime: 2010-08-23 08:27:21
EndTime: 2010-08-23 08:27:21
Duration: 0
BillableSeconds: 0
Disposition: ANSWERED
AMAFlags: DOCUMENTATION
UniqueID: 1282570041.3
UserField: 
Rate: 0.02
Carrier: BS&S

[master]

table = cdr

;
; List the column names to use when inserting CDRs.
;
columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata,
    duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield, 
    test

;
; Map CDR contents to the previously specified columns.
;
values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}',
    '${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}',
    '${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}',
    '${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'

local4.*      /var/log/asterisk/asterisk-cdr.log
[cdr]

facility = local4
priority = info
template = "We received a call from ${CDR(src)}"

$ cat /var/log/asterisk/asterisk-cdr.log

Aug 12 19:17:36 pbx cdr: "We received a call from 2565551212"

exten => *98,1,VoiceMailMain(@${GLOBAL(VOICEMAIL_CONTEXT)})
"""Console"" <2565551212>","2565551212","*98","UserServices","Console/dsp","",
    "VoiceMailMain","@shifteight.org","2010-08-16 01:08:44","2010-08-16 01:08:44",
    "2010-08-16 01:08:53","9","9","ANSWERED","DOCUMENTATION","","1281935324.0","",0

exten => 101,1,Dial(SIP/0000FFFF0002)
"""Console"" <2565551212>","2565551212","101","LocalSets","Console/dsp",
    "SIP/0000FFFF0002-00000000","Dial","SIP/0000FFFF0002","2010-08-16 01:16:10",
    "2010-08-16 01:16:16","2010-08-16 01:16:29","19","13","ANSWERED",
    "DOCUMENTATION","","1281935770.2","",2

exten => 101,1,Set(CHANNEL(userfield)=I like waffles!)
exten => 7,1,CELGenUserEvent(MENU_CHOICE,Caller chose option 7)
*CLI> core show application CELGenUserEvent
[mytable]

connection = asterisk
table = asterisk_cel
alias <CEL field> => <column name>
alias exten => extension
filter <CEL field> => <content>
filter appname => Dial
static <"Static Content Goes Here"> => <column name>
static "My Content" => my_identifier
[mappings]

Master.csv => ${CSV_QUOTE(${eventtype})},${CSV_QUOTE(${eventtime})},
    ${CSV_QUOTE(${CALLERID(name)})},${CSV_QUOTE(${CALLERID(num)})},
    ${CSV_QUOTE(${CALLERID(ANI)})},${CSV_QUOTE(${CALLERID(RDNIS)})},
    ${CSV_QUOTE(${CALLERID(DNID)})},${CSV_QUOTE(${CHANNEL(exten)})},
    ${CSV_QUOTE(${CHANNEL(context)})},${CSV_QUOTE(${CHANNEL(channame)})},
    ${CSV_QUOTE(${CHANNEL(appname)})},${CSV_QUOTE(${CHANNEL(appdata)})},
    ${CSV_QUOTE(${CHANNEL(amaflags)})},${CSV_QUOTE(${CHANNEL(accountcode)})},
    ${CSV_QUOTE(${CHANNEL(uniqueid)})},${CSV_QUOTE(${CHANNEL(linkedid)})},
    ${CSV_QUOTE(${CHANNEL(peer)})},${CSV_QUOTE(${CHANNEL(userfield)})},
    ${CSV_QUOTE(${eventextra})}

[manager]

enabled = yes
exten => 111,1,Answer()
    same => n,CELGenUserEvent(Custom Event,Whiskey Tango Foxtrot)
    same => n,Hangup()

*CLI> console dial 111@testing
Event: CEL
Privilege: call,all
EventName: CHAN_START
AccountCode: 
CallerIDnum: 
CallerIDname: 
CallerIDani: 
CallerIDrdnis: 
CallerIDdnid: 
Exten: 111
Context: testing
Channel: Console/dsp
Application: 
AppData: 
EventTime: 2010-08-23 08:14:51
AMAFlags: NONE
UniqueID: 1282569291.1
LinkedID: 1282569291.1
Userfield: 
Peer: 

[master]

table = cel

;
; List the column names to use when inserting CEL events.
;
columns => eventtype, eventtime, cidname, cidnum, cidani, cidrdnis, ciddnid,
    context, exten, channame, appname, appdata, amaflags, accountcode, uniqueid,
    userfield, peer

;
; Map CEL event contents to the previously specified columns.
;
values => '${eventtype}','${eventtime}','${CALLERID(name)}','${CALLERID(num)}',
    '${CALLERID(ANI)}','${CALLERID(RDNIS)}','${CALLERID(DNID)}',
    '${CHANNEL(context)}','${CHANNEL(exten)}','${CHANNEL(channame)}',
    '${CHANNEL(appname)}','${CHANNEL(appdata)}','${CHANNEL(amaflags)}',
    '${CHANNEL(accountcode)}','${CHANNEL(uniqueid)}','${CHANNEL(userfield)}',
    '${CHANNEL(peer)}'
[general]

enable = yes
apps = Dial,Playback
events = ALL
exten => 200,1,Answer()
    same => n,Playback(hello-world)
    same => n,Hangup()

"CHAN_START","1282062437.436130","Julie Bryant","12565553333","","","","200",
   "LocalSets","SIP/0000FFFF0003-00000010","","","3","","1282062437.17",
   "1282062437.17","",""

"ANSWER","1282062437.436513","Julie Bryant","12565553333","12565553333","",
   "200","200","LocalSets","SIP/0000FFFF0003-00000010","Answer","","3","",
   "1282062437.17","1282062437.17","",""

"APP_START","1282062437.501868","Julie Bryant","12565553333","12565553333",
   "","200","200","LocalSets","SIP/0000FFFF0003-00000010","Playback",
   "hello-world","3","","1282062437.17","1282062437.17","",""

"APP_END","1282062439.008997","Julie Bryant","12565553333","12565553333","",
   "200","200","LocalSets","SIP/0000FFFF0003-00000010","Playback",
   "hello-world","3","","1282062437.17","1282062437.17","",""

"HANGUP","1282062439.009127","Julie Bryant","12565553333","12565553333","",
   "200","200","LocalSets","SIP/0000FFFF0003-00000010","","","3","",
   "1282062437.17","1282062437.17","",""

"CHAN_END","1282062439.009666","Julie Bryant","12565553333","12565553333",
   "","200","200","LocalSets","SIP/0000FFFF0003-00000010","","","3","",
   "1282062437.17","1282062437.17","",""

"LINKEDID_END","1282062439.009707","Julie Bryant","12565553333",
   "12565553333","","200","200","LocalSets","SIP/0000FFFF0003-00000010","",
   "","3","","1282062437.17","1282062437.17","",""
exten => 101,1,Dial(SIP/0000FFFF0001)
"CHAN_START","1282062455.574611","Julie Bryant","12565553333","","","","101",
   "LocalSets","SIP/0000FFFF0003-00000011","","","3","","1282062455.18",
   "1282062455.18","",""

"APP_START","1282062455.574872","Julie Bryant","12565553333","12565553333","",
   "101","101","LocalSets","SIP/0000FFFF0003-00000011","Dial",
   "SIP/0000FFFF0001","3","","1282062455.18","1282062455.18","",""

"CHAN_START","1282062455.575044","Candice Yant","12565551111","","","","s",
   "LocalSets","SIP/0000FFFF0001-00000012","","","3","","1282062455.19",
   "1282062455.18","",""

"ANSWER","1282062458.068134","","101","12565551111","","","101","LocalSets",
   "SIP/0000FFFF0001-00000012","AppDial","(Outgoing Line)","3","",
   "1282062455.19","1282062455.18","",""

"ANSWER","1282062458.068361","Julie Bryant","12565553333","12565553333","",
   "101","101","LocalSets","SIP/0000FFFF0003-00000011","Dial",
   "SIP/0000FFFF0001","3","","1282062455.18","1282062455.18","",""

"BRIDGE_START","1282062458.068388","Julie Bryant","12565553333",
   "12565553333","","101","101","LocalSets","SIP/0000FFFF0003-00000011",
   "Dial","SIP/0000FFFF0001","3","","1282062455.18","1282062455.18","",""

"BRIDGE_END","1282062462.965704","Julie Bryant","12565553333","12565553333",
   "","101","101","LocalSets","SIP/0000FFFF0003-00000011","Dial",
   "SIP/0000FFFF0001","3","","1282062455.18","1282062455.18","",""

"HANGUP","1282062462.966097","","101","12565551111","","","","LocalSets",
   "SIP/0000FFFF0001-00000012","AppDial","(Outgoing Line)","3","",
   "1282062455.19","1282062455.18","",""

"CHAN_END","1282062462.966119","","101","12565551111","","","","LocalSets",
   "SIP/0000FFFF0001-00000012","AppDial","(Outgoing Line)","3","",
   "1282062455.19","1282062455.18","",""

"APP_END","1282062462.966156","Julie Bryant","12565553333","12565553333","",
   "101","101","LocalSets","SIP/0000FFFF0003-00000011","Dial",
   "SIP/0000FFFF0001","3","","1282062455.18","1282062455.18","",""

"HANGUP","1282062462.966215","Julie Bryant","12565553333","12565553333",
   "","101","101","LocalSets","SIP/0000FFFF0003-00000011","","","3","",
   "1282062455.18","1282062455.18","",""

"CHAN_END","1282062462.966418","Julie Bryant","12565553333","12565553333",
   "","101","101","LocalSets","SIP/0000FFFF0003-00000011","","","3","",
   "1282062455.18","1282062455.18","",""

"LINKEDID_END","1282062462.966441","Julie Bryant","12565553333",
   "12565553333","","101","101","LocalSets","SIP/0000FFFF0003-00000011",
   "","","3","","1282062455.18","1282062455.18","",""
exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)
"CHAN_START","1282062488.028200","Julie Bryant","12565553333","","","",
   "102","LocalSets","SIP/0000FFFF0003-00000013","","","3","",
   "1282062488.20","1282062488.20","",""

"APP_START","1282062488.028464","Julie Bryant","12565553333","12565553333",
   "","102","102","LocalSets","SIP/0000FFFF0003-00000013","Dial",
   "SIP/0000FFFF0002","3","","1282062488.20","1282062488.20","",""

"CHAN_START","1282062488.028762","Brooke Brown","12565552222","","","",
   "s","LocalSets","SIP/0000FFFF0002-00000014","","","3","","1282062488.21",
   "1282062488.20","",""

"ANSWER","1282062492.565759","","102","12565552222","","","102","LocalSets",
   "SIP/0000FFFF0002-00000014","AppDial","(Outgoing Line)","3","",
   "1282062488.21","1282062488.20","",""

"ANSWER","1282062492.565973","Julie Bryant","12565553333","12565553333","",
   "102","102","LocalSets","SIP/0000FFFF0003-00000013","Dial",
   "SIP/0000FFFF0002","3","","1282062488.20","1282062488.20","",""

"BRIDGE_START","1282062492.566001","Julie Bryant","12565553333",
   "12565553333","","102","102","LocalSets","SIP/0000FFFF0003-00000013",
   "Dial","SIP/0000FFFF0002","3","","1282062488.20","1282062488.20","",""

"CHAN_START","1282062497.940687","","","","","","s","LocalSets",
   "AsyncGoto/SIP/0000FFFF0002-00000014","","","3","","1282062497.22",
   "1282062488.20","",""

"BLINDTRANSFER","1282062497.940925","Julie Bryant","12565553333","12565553333","",
   "102","102","LocalSets","SIP/0000FFFF0003-00000013","Dial","SIP/0000FFFF0002",
   "3","","1282062488.20","1282062488.20",
   "AsyncGoto/SIP/0000FFFF0002-00000014<ZOMBIE>",""

"BRIDGE_END","1282062497.940961","Julie Bryant","12565553333","12565553333","",
   "102","102","LocalSets","SIP/0000FFFF0003-00000013","Dial",
   "SIP/0000FFFF0002","3","","1282062488.20","1282062488.20","",""

"APP_START","1282062497.941021","","102","12565552222","","","101","LocalSets",
   "SIP/0000FFFF0002-00000014","Dial","SIP/0000FFFF0001","3","",
   "1282062497.22","1282062488.20","",""

"CHAN_START","1282062497.941207","Candice Yant","12565551111","","","","s",
   "LocalSets","SIP/0000FFFF0001-00000015","","","3","","1282062497.23",
   "1282062488.20","",""

"HANGUP","1282062497.941361","","","","","","","LocalSets",
   "AsyncGoto/SIP/0000FFFF0002-00000014<ZOMBIE>","AppDial",
   "(Outgoing Line)","3","","1282062488.21","1282062488.20","",""

"CHAN_END","1282062497.941380","","","","","","","LocalSets",
   "AsyncGoto/SIP/0000FFFF0002-00000014<ZOMBIE>","AppDial","(Outgoing Line)",
   "3","","1282062488.21","1282062488.20","",""

"APP_END","1282062497.941415","Julie Bryant","12565553333","12565553333","",
   "102","102","LocalSets","SIP/0000FFFF0003-00000013","Dial",
   "SIP/0000FFFF0002","3","","1282062488.20","1282062488.20","",""

"HANGUP","1282062497.941453","Julie Bryant","12565553333","12565553333",
   "","102","102","LocalSets","SIP/0000FFFF0003-00000013","","","3","",
   "1282062488.20","1282062488.20","",""

"CHAN_END","1282062497.941474","Julie Bryant","12565553333","12565553333",
   "","102","102","LocalSets","SIP/0000FFFF0003-00000013","","","3","",
   "1282062488.20","1282062488.20","",""

"ANSWER","1282062500.559578","","101","12565551111","","","101","LocalSets",
   "SIP/0000FFFF0001-00000015","AppDial","(Outgoing Line)","3","",
   "1282062497.23","1282062488.20","",""

"BRIDGE_START","1282062500.559720","","102","12565552222","","","101","LocalSets",
   "SIP/0000FFFF0002-00000014","Dial","SIP/0000FFFF0001","3","","1282062497.22",
   "1282062488.20","",""

"BRIDGE_END","1282062512.742600","","102","12565552222","","","101","LocalSets",
   "SIP/0000FFFF0002-00000014","Dial","SIP/0000FFFF0001","3","","1282062497.22",
   "1282062488.20","",""

"HANGUP","1282062512.743006","","101","12565551111","","","","LocalSets",
   "SIP/0000FFFF0001-00000015","AppDial","(Outgoing Line)","3","","1282062497.23",
   "1282062488.20","",""

"CHAN_END","1282062512.743211","","101","12565551111","","","","LocalSets",
   "SIP/0000FFFF0001-00000015","AppDial","(Outgoing Line)","3","","1282062497.23",
   "1282062488.20","",""

"APP_END","1282062512.743286","","102","12565552222","","","101","LocalSets",
   "SIP/0000FFFF0002-00000014","Dial","SIP/0000FFFF0001","3","","1282062497.22",
   "1282062488.20","",""

"HANGUP","1282062512.743346","","102","12565552222","","","101","LocalSets",
   "SIP/0000FFFF0002-00000014","","","3","","1282062497.22","1282062488.20",
   "",""

"CHAN_END","1282062512.743371","","102","12565552222","","","101","LocalSets",
   "SIP/0000FFFF0002-00000014","","","3","","1282062497.22","1282062488.20",
   "",""

"LINKEDID_END","1282062512.743391","","102","12565552222","","","101",
   "LocalSets","SIP/0000FFFF0002-00000014","","","3","","1282062497.22",
   "1282062488.20","",""

$ sudo yum install net-snmp-devel
$ sudo apt-get install snmp libsnmp-dev snmpd
$ cd ~/src/asterisk-complete/asterisk/1.8/
$ ./configure
$ make menuselect  # verify that res_snmp is selected under Resource Modules 
$ sudo make install

$ sudo cp ~/src/asterisk-complete/asterisk/1.8/configs/res_snmp.conf.sample \ 
/etc/asterisk/res_snmp.conf
[general]
;subagent=yes
;enabled=yes
[general]
subagent=yes
enabled=yes
*CLI> module unload res_snmp.so

Unloaded res_snmp.so
 Unloading [Sub]Agent Module
  == Terminating SubAgent

*CLI> module load res_snmp.so

Loaded res_snmp.so
  == Parsing '/etc/asterisk/res_snmp.conf':   == Found
 Loading [Sub]Agent Module
 Loaded res_snmp.so => (SNMP [Sub]Agent for Asterisk)
  == Starting SubAgent
$ cd /etc/snmp
$ sudo mv snmpd.conf snmpd.sample
$ sudo cat > snmpd.conf

com2sec notConfigUser  default       public

group   notConfigGroup v1           notConfigUser
group   notConfigGroup v2c           notConfigUser

view all    included  .1
view system included  .iso.org.dod.internet.mgmt.mib-2.system

access  notConfigGroup ""      any       noauth    exact  all    none none

syslocation Caledon, ON
syscontact Leif Madsen lmadsen@shifteight.org

Ctrl+D
$ sudo cat >> snmpd.conf

master agentx
agentXSocket /var/agentx/master
agentXPerms 0660 0775 nobody root

sysObjectID .1.3.6.1.4.1.22736.1

Ctrl+D
$ sudo /etc/init.d/snmpd restart
$ snmpwalk -On -v2c -c public 127.0.0.1 .1.3.6.1.4.1.22736
.1.3.6.1.4.1.22736.1.5.4.1.4.3 = INTEGER: 2
.1.3.6.1.4.1.22736.1.5.4.1.4.4 = INTEGER: 2
.1.3.6.1.4.1.22736.1.5.4.1.4.5 = INTEGER: 1
.1.3.6.1.4.1.22736.1.5.4.1.4.6 = INTEGER: 1
.1.3.6.1.4.1.22736.1.5.4.1.5.1 = INTEGER: 1
...etc
$ sudo /etc/init.d/snmpd stop
createUser initial MD5 setup_passphrase DES
rwuser initial
$ sudo /etc/init.d/snmpd start
Starting snmpd:                                            [  OK  ]
$ sudo snmpusm -v3 -u initial -n "" -l authPriv \
-a MD5 -A setup_passphrase \
-x DES -X setup_passphrase \
localhost create opennmsUser initial
User successfully created.
$ sudo snmpusm -v 3 -u initial -n "" -l authPriv \
-a MD5 -A setup_passphrase \
-x DES -X setup_passphrase \
-Ca -Cx localhost passwd setup_passphrase \
0p3nNMSv3 opennmsUser
SNMPv3 Key(s) successfully changed.
$ sudo snmpwalk -v 3 -u opennmsUser -n "" -l authPriv \
-a MD5 -A 0p3nNMSv3 \
-x DES -X 0p3nNMSv3 \
localhost ifTable
IF-MIB::ifIndex.1 = INTEGER: 1
IF-MIB::ifIndex.2 = INTEGER: 2
IF-MIB::ifIndex.3 = INTEGER: 3
IF-MIB::ifDescr.1 = STRING: lo
IF-MIB::ifDescr.2 = STRING: eth0
IF-MIB::ifDescr.3 = STRING: sit0

com2sec notConfigUser  default       public

#group   notConfigGroup v1            notConfigUser
#group   notConfigGroup v2c           notConfigUser
group   notConfigGroup usm           opennmsUser

#view    systemview    included   .1.3.6.1.2.1.1
#view    systemview    included   .1.3.6.1.2.1.25.1.1
view    all           included   .1

#access  notConfigGroup ""      any       noauth    exact  all    none none
access  notConfigGroup ""       usm       priv      exact  all    none none

syslocation Caledon, ON
syscontact Leif Madsen lmadsen@shifteight.org

#rwuser initial

master agentx
agentXSocket /var/agentx/master
agentXPerms 0660 0775 nobody root

sysObjectID .1.3.6.1.4.1.22736.1

$ sudo /etc/init.d/snmpd restart
$ sudo snmpwalk -v 3 -u opennmsUser -n "" -l authPriv \
-a MD5 -A 0p3nNMSv3 \
-x DES -X 0p3nNMSv3 \
localhost ifTable
$ sudo snmpwalk -v 3 -u opennmsUser -n "" -l authPriv \
-a MD5 -A 0p3nNMSv3 \
-x DES -X 0p3nNMSv3 \
localhost .1.3.6.1.4.1.22736
SNMPv2-SMI::enterprises.22736.1.5.4.1.2.1 = STRING: "SIP"
SNMPv2-SMI::enterprises.22736.1.5.4.1.2.2 = STRING: "IAX2"
SNMPv2-SMI::enterprises.22736.1.5.4.1.2.3 = STRING: "Bridge"
SNMPv2-SMI::enterprises.22736.1.5.4.1.2.4 = STRING: "MulticastRTP"
SNMPv2-SMI::enterprises.22736.1.5.4.1.2.5 = STRING: "DAHDI"
SNMPv2-SMI::enterprises.22736.1.5.4.1.2.6 = STRING: "Local"
SNMPv2-SMI::enterprises.22736.1.5.4.1.3.1 = STRING: "Session Initiation Protocol(SIP)"
SNMPv2-SMI::enterprises.22736.1.5.4.1.3.2 = STRING: "Inter Asterisk eXchange Driver"
SNMPv2-SMI::enterprises.22736.1.5.4.1.3.3 = STRING: "Bridge Interaction Channel"
SNMPv2-SMI::enterprises.22736.1.5.4.1.3.4 = STRING: "Multicast RTP Paging Channel"
SNMPv2-SMI::enterprises.22736.1.5.4.1.3.5 = STRING: "DAHDI Telephony Driver"
SNMPv2-SMI::enterprises.22736.1.5.4.1.3.6 = STRING: "Local Proxy Channel Driver"
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"123"<sip:123@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"1234"<sip:1234@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"12345"<sip:12345@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"123456"<sip:123456@127.0.0.1>' failed for '203.86.167.220:5061' - No matching 
peer found
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"test"<sip:test@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"sip"<sip:sip@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found
[Aug 22 15:17:15] NOTICE[25690] chan_sip.c: Registration from 
'"user"<sip:user@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"admin"<sip:admin@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"pass"<sip:pass@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"password"<sip:password@127.0.0.1>' failed for '203.86.167.220:5061' - No matching 
peer found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"testing"<sip:testing@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"guest"<sip:guest@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"voip"<sip:voip@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer 
found
[Aug 22 15:17:16] NOTICE[25690] chan_sip.c: Registration from 
'"account"<sip:account@127.0.0.1>' failed for '203.86.167.220:5061' - No matching 
peer found

...

[Aug 22 15:17:17] NOTICE[25690] chan_sip.c: Registration from 
'"100"<sip:100@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found
[Aug 22 15:17:17] NOTICE[25690] chan_sip.c: Registration from 
'"101"<sip:101@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found
[Aug 22 15:17:17] NOTICE[25690] chan_sip.c: Registration from 
'"102"<sip:102@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found
[Aug 22 15:17:17] NOTICE[25690] chan_sip.c: Registration from 
'"103"<sip:103@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found
[Aug 22 15:17:17] NOTICE[25690] chan_sip.c: Registration from 
'"104"<sip:104@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found
[Aug 22 15:17:17] NOTICE[25690] chan_sip.c: Registration from 
'"105"<sip:105@127.0.0.1>' failed for '203.86.167.220:5061' - No matching peer found

$ sudo apt-get install fail2ban
$ sudo yum install fail2ban
$ sudo apt-get install iptables
$ sudo yum install iptables
$ sudo iptables -L

Chain INPUT (policy ACCEPT)
target     prot opt source               destination         

Chain FORWARD (policy ACCEPT)
target     prot opt source               destination         

Chain OUTPUT (policy ACCEPT)
target     prot opt source               destination        
          $
           sudo apt-get install postfix

$ sudo yum install postfix
$ echo "Just testing." > email.txt
$ mutt -s "Testing" youraddress@shifteight.org < email.txt
[general]

dateformat = %F %T

[logfiles]

console => notice,warning,error,debug
messages => notice,warning,error

[INCLUDES]

# Read common prefixes. If any customizations available -- read them from
# common.local
#before = common.conf

[Definition]

#_daemon = asterisk

# Option:  failregex
# Notes.:  regex to match the password failures messages in the logfile. The
#          host must be matched by a group named "host". The tag "<HOST>" can
#          be used for standard IP/hostname matching and is only an alias for
#          (?:::f{4,6}:)?(?P<host>\S+)
# Values:  TEXT
#

# *** All lines below should start with NOTICE
#     Some lines have been wrapped due to space requirements for
#     the book. All new lines should start with NOTICE.
#
failregex = NOTICE.* .*: Registration from '.*' failed for '<HOST>' 
 - Wrong password
            NOTICE.* .*: Registration from '.*' failed for '<HOST>' 
 - No matching peer found
            NOTICE.* .*: Registration from '.*' failed for '<HOST>' 
 - Username/auth name mismatch
            NOTICE.* .*: Registration from '.*' failed for '<HOST>' 
 - Device does not match ACL
            NOTICE.* <HOST> failed to authenticate as '.*'$
            NOTICE.* .*: No registration for peer '.*' \(from <HOST>\)
            NOTICE.* .*: Host <HOST> failed MD5 authentication for '.*' (.*)
            NOTICE.* .*: Failed to authenticate user .*@<HOST>.*

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =

[asterisk-iptables]

enabled  = true
filter   = asterisk
action   = iptables-allports[name=ASTERISK, protocol=all]
           sendmail-whois[name=ASTERISK, dest=me@shifteight.org, 
           sender=fail2ban@shifteight.org]
logpath  = /var/log/asterisk/messages
maxretry = 5
bantime = 259200

[DEFAULT]

# Multiple addresses can be specified, separated by a space.
ignoreip = 127.0.0.1 10.1.1.1

destemail = youraddress@shifteight.org

exten => _X.,1,Dial(IAX2/otherserver/${EXTEN},30)
1234&DAHDI/g1/12565551212
exten => _X.,1,Dial(IAX2/otherserver/1234&DAHDI/g1/12565551212,30)
exten => _XXXX,1,Dial(IAX2/otherserver/${EXTEN},30)
exten => _X.,1,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})})
    same => n,Dial(IAX2/otherserver/${SAFE_EXTEN},30)
[general]

calltokenoptional = 0.0.0.0/0.0.0.0 
maxcallnumbers = 16382

[general]

; Set the default call number limit per host
maxcallnumbers = 16

[callnumberlimits]

; Set a different call number limit for all hosts in a
; specified range.
192.168.1.0/255.255.255.0 = 1024

[some_peer]

; A dynamic peer's address is not known until that peer
; registers. A call number limit can be specified in the
; peer's section instead of the callnumberlimits section.

type = peer
host = dynamic
maxcallnumbers = 512
[some_other_peer]

requirecalltoken = auto
[general]

maxcallnumbers_nonvalidated = 2048

[guest]

type = user
requirecalltoken = no
[phoneA] ; Use a better account name than this.

type = friend

; Start by denying everyone.
deny = 0.0.0.0/0.0.0.0

; Allow connections that originate from 192.168.X.X to attempt
; to authenticate against this account.
permit = 192.168.0.0/255.255.0.0
exten => _X.,1,Set(GROUP(users)=${CHANNEL(peername)})

; *** This line should have no line breaks
    same => n,NoOp(There are ${GROUP_COUNT(${CHANNEL(peername)})} 
calls for account ${CHANNEL(peername)}.)

    same => n,GotoIf($[${GROUP_COUNT(${CHANNEL(peername)})} > 2]?denied:continue)
    same => n(denied),NoOp(There are too many calls up already.  Hang up.)
    same => n,HangUp()
    same => n(continue),NoOp(continue processing call as normal here ...)

0011 0101 0100 1001 1011 1011 1010 0001 0101 0101 0000 1100 1100 1010
00111 01000 01001 01001 01000 00101 10110 11000 11001 11001 11000 10111 
10100 10001 00010 00111 01001 01010 01001 00111 00000 11000 11010 11010 
11001 11000 10110 10001
00000 10011 10100 10101 01101 00001 00011 11010 00010 00001 01000 10011 
10100 10100 00101 00100 00101 10101 10011 10001 00011 00001 00000 10100 
10010 10101 01101 10100 00101 11010 00100 00000 01000
register => username:secret@my_remote_peer

Name	Purpose	Popularity/Status
`app_adsiprog`	Loads Analog Display Services Interface (ADSI) scripts into compatible analog phones	Insignificant
`app_alarmreceiver`	Supports receipt of reports from alarm equipment	Insignificant
`app_amd`	Detects answering machines	Unreliable
`app_authenticate`	Compares dual-tone multi-frequency (DTMF) input against a provided string (password)	Useful
`app_cdr`	Writes ad hoc record to CDR	Useful
`app_celgenuserevent`	Generates user-defined events for CEL	New
`app_chanisavail`	Checks the status of a channel	Unreliable
`app_channelredirect`	Forces another channel into a different part of the dialplan	Useful
`app_chanspy`	Allows a channel to listen to audio on another channel	Useful
`app_confbridge`	Provides conferencing (new version)	New—not fully featured yet
`app_controlplayback`	Plays back a prompt and offers fast forward and rewind functions	Useful
`app_dahdibarge`	Allows barging in on a DAHDI channel	Deprecated—see `app_chanspy`
`app_dahdiras`	Creates a RAS server over a DAHDI channel (no modem emulation)	Insignificant
`app_db`	Used to add/change/delete records in Asterisk’s built-in Berkeley database	Deprecated—see `func_db`
`app_dial`	Used to connect channels together (i.e., make phone calls)	Essential
`app_dictate`	Plays back a recording and offers start/stop functions	Useful
`app_directed_pickup`	Answers a call for another extension	Useful
`app_directory`	Presents the list of names from `voicemail.conf`	Useful
`app_disa`	Provides dialtone and accepts DTMF input	Useful^[a]
`app_dumpchan`	Dumps channel variables to Asterisk command-line interface (CLI)	Useful
`app_echo`	Loops received audio back to source channel	Useful
`app_exec`	Contains `Exec()`, `TryExec()`, and `ExecIf()`; executes a dialplan application based on conditions	Useful
`app_externalivr`	Controls Asterisk as with an AGI, only asynchronously	Useful
`app_fax`	Provides `SendFax()` and `ReceiveFax()`	Useful^[b]
`app_festival`	Enables basic text to speech using Festival TTS engine	Usable
`app_flash`	Performs a hook-switch flash on channels (primarily analog)	Useful
`app_followme`	Performs find me/follow me functionality based on `followme.conf`	Useful
`app_forkcdr`	Starts new CDR record on current call	Usable
`app_getcpeid`	Gets the ADSI CPE ID	Insignificant
`app_ices`	Sends audio to an Icecast server	Usable
`app_image`	Transmits an image to supported devices	Limited
`app_ivrdemo`	Sample application for developers	Insignificant
`app_jack`	Works with JACK Audio Connection Kit to share audio between compatible applications	Useful
`app_macro`	Triggers dialplan macros	Deprecated—see `GoSub()`
`app_meetme`	Provides multiparty conferencing	Useful—fully featured
`app_milliwatt`	Generates 1004-Hz tone for testing loss on analog circuits	Useful
`app_minivm`	Provides primitive functions to allow you to build your own voicemail application in dialplan	Usable
`app_mixmonitor`	Records both sides of a call and mixes them together	Useful
`app_morsecode`	Generates Morse code	Usable
`app_mp3`	Uses mpg123 to play an MP3	Insignificant
`app_nbscat`	Streams audio from Network Broadcast Stream (NBS)	Insignificant
`app_originate`	Allows origination of a call	Useful
`app_osplookup`	Performs Open Settlement Protocol (OSP) lookup	Usable
`app_page`	Creates multiple audio connections to specified devices for public address (paging)	Useful
`app_parkandannounce`	Enables automated announcing of parked calls	Usable
`app_playback`	Plays a file to the channel (does not accept input)	Useful
`app_playtones`	Plays pairs of tones of specified frequencies	Useful
`app_privacy`	Requests input of caller’s phone number if no CallerID is received	Insignificant
`app_queue`	Provides Automatic Call Distribution (ACD)	Useful
`app_read`	Requests input of digits from callers and assigns input to a variable	Useful
`app_readexten`	Requests input of digits from callers and passes call to a designated extension and context	Usable
`app_readfile`	Loads contents of a text file into a channel variable	Deprecated—see the `FILE()` function in `func_env`
`app_record`	Records received audio to a file	Useful
`app_rpt`	Provides a method to interface with an audio board for the `app_rpt` project	Limited
`app_sayunixtime`	Plays back time in specified format	Useful
`app_senddtmf`	Transmits DTMF to calling party	Useful
`app_sendtext`	Sends a text string to compatible channels	Insignificant
`app_setcallerid`	Sets CallerID on a channel	Deprecated—see `func_callerid`
`app_skel`	Sample application for developers	Useful^[c]
`app_sms`	Sends SMS message in compatible countries	Limited
`app_softhangup`	Requests hangup of channel	Useful
`app_speech_utils`	Provides utilities relating to speech recognition	Useful^[d]
`app_stack`	Provides `Gosub()`, `GoSubIf()`, `Return()`, `StackPop()`, `LOCAL()`, and `LOCAL_PEEK()`	Essential
`app_system`	Executes commands in a Linux shell	Useful
`app_talkdetect`	Similar to `app_background`, but allows for any received audio to interrupt playback	Useful
`app_test`	Client/server testing application	Usable
`app_transfer`	Performs a transfer on the current channel	Useful
`app_url`	Passes a URI to the called channel	Limited
`app_userevent`	Generates a custom event in the Asterisk Manager Interface (AMI)	Useful
`app_verbose`	Generates a custom event in the Asterisk CLI	Useful
`app_voicemail`	Provides voicemail	Essential
`app_waitforring`	Waits for a RING signaling event (not to be confused with RINGING); most likely unnecessary, as only `chan_dahdi` with analog channels where ringing is received (such as an FXO port) generates the RING signaling event	Insignificant
`app_waitforsilence`	Includes `WaitForSilence()` and `WaitForNoise()`; listens to the incoming channel for a specified number of milliseconds of noise/silence	Useful
`app_waituntil`	Waits until current Linux epoch matches specified epoch	Useful
`app_while`	Includes `While()`, `EndWhile()`, `ExitWhile()`, and `ContinueWhile()`; provides while-loop functionality in the dialplan	Useful
`app_zapateller`	Generates SIT tone to discourage telemarketers	Usable
^[a]The use of (DISA) is considered to be a security risk. ^[b]Requires a suitable DSP engine to handle encoding/decoding of fax signaling (see Chapter 19, Fax). ^[c]If you are a developer. ^[d]Requires an external speech recognition application.

Name	Purpose	Popularity/Status
`bridge_builtin_features`	Performs bridging when utilizing built-in user features (such as those found in `features.conf`).	New
`bridge_multiplexed`	Performs complex multiplexing, as would be required in a large conference room (multiple participants). Currently only used by `app_confbridge`.	New
`bridge_simple`	Performs simple channel-to-channel bridging.	New
`bridge_softmix`	Performs simple multiplexing, as would be required in a large conference room (multiple participants). Currently only used by `app_confbridge`.	New

Name	Purpose	Popularity/Status
`cdr_adaptive_odbc`	Allows writing of CDRs through ODBC framework with ability to add custom fields	Useful
`cdr_csv`	Writes CDRs to disk as a comma-separated values file	Usable
`cdr_custom`	As above, but allows for the addition of custom fields	Useful
`cdr_manager`	Outputs CDRs to Asterisk Manager Interface (AMI)	Useful
`cdr_odbc`	Writes CDRs through ODBC framework	Usable
`cdr_pgsql`	Writes CDRs to PostgreSQL	Useful
`cdr_radius`	Writes CDRs to RADIUS	Usable—does not support custom fields
`cdr_sqlite`	Writes CDRs to SQLite2 database	Deprecated—use `sqlite3_custom`
`cdr_sqlite3_custom`	Writes CDRs to SQLite3 with custom fields	Useful
`cdr_syslog`	Writes CDRs to syslog	Useful
`cdr_tds`	Writes CDRs to Microsoft SQL or Sybase	Usable—requires an old version of `libtds`

Name	Purpose	Popularity/Status
`cel_custom`	CEL to disk/file	Useful
`cel_manager`	CEL to AMI	Useful
`cel_odbc`	CEL to ODBC	Useful
`cel_pgsql`	CEL to PostgreSQL	Useful
`cel_radius`	CEL to RADIUS	Usable—does not support custom fields
`cel_sqlite3_custom`	CEL to Sqlite3	Useful
`cel_tds`	CEL to Microsoft SQL or Sybase	Usable—requires an old version of `libtds`

Name	Purpose	Popularity/Status
`chan_agent`	Provides agent channel for `Queue()`	Useful
`\|chan_alsa`	Provides connection to Advanced Linux Sound Architecture	Useful
`chan_bridge`	Used internally by the `ConfBridge()` application; should not be used directly	Essential ^[a]
`chan_console`	Provides connection to portaudio	New
`chan_dahdi`	Provides connection to PSTN cards that use DAHDI channel drivers	Useful
`chan_gtalk`	Provides connection to Google Talk	Usable
`chan_h323`	Provides connection to H.323 endpoints	Deprecated—see `chan_ooh323` in Table 2.11, “Addon modules”
`chan_iax2`	Provides connection to IAX2 endpoints	Useful
`chan_jingle`	Provides connection to Jingle-enabled endpoints	Usable
`chan_local`	Provides a mechanism to treat a portion of the dialplan as a channel	Useful
`chan_mgcp`	Media Gateway Control Protocol channel driver	Usable
`chan_misdn`	Provides connection to mISDN supported ISDN cards	Limited
`chan_multicast_rtp`	Provides connection to multicast RTP streams	Useful
`chan_nbs`	Network Broadcast Sound channel driver	Insignificant
`chan_oss`	Open Sound System driver	Useful
`chan_phone`	Linux telephony interface driver (very old)	Insignificant
`chan_sip`	Session Initiation Protocol channel driver	Essential
`chan_skinny`	Cisco Skinny Client Control Protocol (SCCP) channel driver	Usable
`chan_unistim`	Nortel Unistim protocol channel driver	Usable
`chan_usbradio`	Channel driver for CM108 USB cards with radio interface	Usable
`chan_vpb`	Voicetronix channel driver	Insignificant^[b]
^[a]If you are using the `ConfBridge()` application. ^[b]Some Voicetronix hardware is supported by Zaptel using an addon Zaptel module distributed by Voicetronix. However, Zaptel is no longer supported by Asterisk and this driver has not been ported to DAHDI.

Name	Purpose	Popularity/Status
`codec_adpcm`	Adaptive Differential Pulse Coded Modulation codec	Insignificant
`codec_alaw`	A-law PCM codec used all over the world (except Canada/USA) on the PSTN	Essential
`codec_a_mu`	A-law to mu-law direct converter	Useful
`codec_dahdi`	Utilizes proprietary Digium hardware transcoding card	Essential^[a]
`codec_g722`	Wideband audio codec	Useful
`codec_g726`	Flavor of ADPCM	Insignificant
`codec_gsm`	Global System for Mobile Communications (GSM) codec	Useful
`codec_ilbc`	Internet Low Bitrate Codec	Insignificant
`codec_lpc10`	Linear Predictive Coding vocoder (extremely low bandwidth)	Insignificant
`codec_resample`	Resamples between 8-bit and 16-bit signed linear	Usable
`codec_speex`	Speex codec	Usable
`codec_ulaw`	Mu-law PCM codec used in Canada/USA on PSTN	Essential
^[a]If you are using a Digium codec transcoder card.

Name	Plays files stored in	Popularity/Status
`format_g723`	G.723 `.g723`	Insignificant
`format_g726`	G.726 `.g726`	Insignificant
`format_g729`	G.729 `.g729`	Useful
`format_gsm`	RPE-LTP (original GSM codec) `.gsm`	Usable
`format_h263`	H.263—video `.h263`	Usable
`format_h264`	H.264—video `.h264`	Usable
`format_ilbc`	Internet Low Bitrate Codec `.ilbc`	Insignificant
`format_jpeg`	Graphic file `.jpeg` `.jpg`	Insignificant
`format` `_ogg_` `vorbis`	Ogg container `.ogg`	Usable
`format_pcm`	Various Pulse-Coded Modulation formats: `.alaw`, `.al`, `.alw`, `.pcm`, `.ulaw`, `.ul`, `.mu`, `.ulw`, `.g722`, `.au`	Useful
`format_siren14`	G.722.1 Annex C (14 kHz) `.siren14`	New
`format_siren7`	G.722.1 (7 kHz) `.siren7`	New
`format_sln16`	16-bit signed linear `.sln16`	New
`format_sln`	8-bit signed linear `.sln` `.raw`	Useful
`format_vox`	`.vox`	Insignificant
`format_wav`	`.wav`	Useful
`format_wav_gsm`	GSM audio in a WAV container `.WAV`, `.wav49`	Usable

Name	Purpose	Popularity/Status
`func_aes`	Encrypts/decrypts an AES string	Useful
`func_audiohookinherit`	Allows calls to be recorded after transfer	Useful
`func_base64`	Encodes/decodes a base-64 string	Usable
`func_blacklist`	Writes/reads blacklist in `astdb`	Useful
`func_callcompletion`	Gets/sets call completion configuration parameters for the channel	New
`func_callerid`	Gets/sets CallerID	Useful
`func_cdr`	Gets/sets CDR variable	Useful
`func_channel`	Gets/sets channel information	Useful
`func_config`	Includes `AST_CONFIG()`; reads variables from config file	Usable
`func_connectedline`	Changes connected line information on supported handsets	New
`func_curl`	Uses cURL to obtain data from a URI	Useful
`func_cut`	Slices and dices strings	Useful
`func_db`	Provides `astdb` functions	Useful
`func_devstate`	Gets state of device	Useful
`func_dialgroup`	Creates a group for simultaneous dialing	Useful
`func_dialplan`	Validates that designated target exists in dialplan	Useful
`func_enum`	Performs ENUM lookup	Useful
`func_env`	Includes `FILE()`, `STAT()`, and `ENV()`; performs operating system actions	Useful
`func_extstate`	Returns status of a hinted extension	Useful
`func_global`	Gets/sets global variables	Useful
`func_groupcount`	Gets/sets channel count for members of a group	Useful
`func_iconv`	Converts between character sets	Usable
`func_lock`	Includes `LOCK()`, `UNLOCK()`, and `TRYLOCK()`; sets a lock that can be used to avoid race conditions in the dialplan	Useful
`func_logic`	Includes `ISNULL()`, `SET()`, `EXISTS()`, `IF()`, `IFTIME()`, and `IMPORT()`; performs various logical functions	Useful
`func_math`	Includes `MATH()`, `INC()`, and `DEC()`; performs mathematical functions	Useful
`func_md5`	Converts supplied string to an MD5 hash	Useful
`func_module`	Checks to see if supplied module is loaded into memory	Usable
`func_odbc`	Allows dialplan integration with ODBC resources	Useful
`func_pitchshift`	Shifts the pitch of an audio stream	Useful
`func_rand`	Returns a random number within a given range	Useful
`func_realtime`	Performs lookups within the Asterisk Realtime Architecture (ARA)	Useful
`func_redirecting`	Provides access to information about where this call was redirected from	Useful
`func_sha1`	Converts supplied string to an SHA1 hash	Useful
`func_shell`	Performs Linux shell operations and returns results	Useful
`func_speex`	Reduces noise and performs dB gain/loss on an audio stream	Useful
`func_sprintf`	Performs string format functions similar to C function of same name	Useful
`func_srv`	Perform SRV lookups in the dialplan	Useful
`func_strings`	Includes over a dozen string manipulation functions	Useful
`func_sysinfo`	Gets system information such as RAM, swap, load average, etc.	Useful
`func_timeout`	Gets/sets timeouts on channel	Useful
`func_uri`	Converts strings to URI-safe encoding	Useful
`func_version`	Returns Asterisk version information	Usable
`func_vmcount`	Returns count of messages in a voicemail folder for a particular user	Useful
`func_volume`	Sets volume on a channel	Useful

Name	Purpose	Popularity/Status
`pbx_ael`	Asterisk Extension Logic (AEL) offers a dialplan scripting language that looks like a modern programming language.	Usable^[a]
`pbx_config`	This is the traditional, and most popular, dialplan language for Asterisk. Without this module, Asterisk cannot read `extensions.conf`.	Useful
`pbx_dundi`	Performs data lookups on remote Asterisk systems.	Useful
`pbx_loopback`	Performs something similar to a dialplan include, but in a deprecated manner.	Insignificant^[b]
`pbx_lua`	Allows creation of a dialplan using the Lua scripting language.	Useful
`pbx_realtime`	Provides functionality related to the Asterisk Realtime Architecture.	Useful
`pbx_spool`	Provides outgoing spool support relating to Asterisk call files.	Useful
^[a]We have not found too many people using AEL. We suspect this is because most developers will tend to use AGI/AMI if they do not want to use traditional dialplans. ^[b]We’ve never heard of this being used in production.

Name	Purpose	Popularity/Status
`res_adsi`	Provides ADSI	Essential^[a]
`res_ael_share`	Provides shared routines for use with `pbx_ael`	Essential if you’re using AEL
`res_agi`	Provides Asterisk Gateway Interface	Useful
`res_ais`	Provides distributed message waiting indication (MWI) and device state notifications via an implementation of the AIS standard, such as OpenAIS	Useful
`res_calendar`	Enables base integration to calendaring systems	Useful
`res_calendar_caldav`	Provides CalDAV-specific capabilities	Useful
`res_calendar_exchange`	Provides MS Exchange capabilities	Useful
`res_calendar_icalendar`	Provides Apple/Google iCalendar capabilities	Useful
`res_clialiases`	Creates CLI aliases	Useful
`res_clioriginate`	Originates a call from the CLI	Usable
`res_config_curl`	Pulls configuration information using cURL	Useful
`res_config_ldap`	Pulls configuration information using LDAP	Usable
`res_config_odbc`	Pulls configuration information using ODBC	Useful
`res_config_pgsql`	Pulls configuration information using PostgreSQL	Usable
`res_config_sqlite`	Pulls configuration information using SQLite	Usable
`res_convert`	Uses the CLI to perform file conversions	Usable
`res_crypto`	Provides cryptographic capabilities	Useful
`res_curl`	Provides common subroutines for other cURL modules	Useful
`res_fax`	Provides common subroutines for other fax modules	Useful
`res_fax_spandsp`	Plug-in for fax using the `spandsp` package	Useful
`res_http_post`	Provides POST upload support for the Asterisk HTTP server	Usable
`res_jabber`	Provides Jabber/XMPP resources	Useful
`res_limit`	Enables adjusting of system limits on the Asterisk process	Usable
`res_monitor`	Provides call recording resources	Useful
`res_musiconhold`	Provides music on hold (MOH) resources	Essential
`res_mutestream`	Allows muting/unmuting of audio streams	New
`res_odbc`	Provides common subroutines for other ODBC modules	Useful
`res_phoneprov`	Provisions phones from Asterisk HTTP server	New
`res_pktccops`	Provides PacketCable COPS resources	New
`res_realtime`	Provides CLI commands for the Asterisk Realtime Architecture (ARA)	Useful
`res_rtp_asterisk`	Provides RTP	Essential
`res_rtp_multicast`	Provides multicast-RTP	New
`res_security_log`	Enables security logging	New
`res_smdi`	Provides voicemail notification using the SMDI protocol	Limited
`res_snmp`	Provides system status information to an SNMP-managed network	Usable
`res_speech`	Generic speech recognition API	Limited^[b]
`res_timing_dahdi`	Provides timing using the DAHDI kernel interface	Useful
`res_timing_kqueue`	Provides timing using a kernel feature in some BSDs, including Mac OS X	New
`res_timing_pthread`	Provides timing using only parts of the standard `pthread` API; less efficient but more portable than other timing modules.	Useful
`res_timing_timerfd`	Provides timing using the `timerfd` API provided by newer versions of the Linux kernel	Useful
^[a]While most of the ADSI functionality in Asterisk is never used, the voicemail application uses this resource. ^[b]Requires a separately licensed product in order to be used.

Name	Purpose	Popularity/Status
`app_mysql`	Executes MySQL queries with a dialplan application	Deprecated—see `func_odbc`
`app_saycountpl`	Says Polish counting words	Deprecated—now integrated in `say.conf`
`cdr_mysql`	Logs call detail records to a MySQL database	Usable—we recommend `cdr_adaptive_odbc` instead
`chan_mobile`	Enables making and receiving phone calls using cell phones over Bluetooth	Limited^[a]
`chan_ooh323`	Enables making and receiving VoIP calls using the H.323 protocol	Usable
`format_mp3`	Allows Asterisk to play MP3 files	Usable
`res_config_mysql`	Uses a MySQL database as a real-time configuration backend	Useful
^[a]While `chan_mobile` works great with many phones, problems have been reported with some models. When a problem does occur, it is very difficult for developers to solve unless they have a phone of the same model to test with.

CentOS	`yum update -y && reboot`
CentOS 64-bit	*`yum remove .i386 && yum update -y && reboot`**
Ubuntu	`sudo apt-get update && sudo apt-get upgrade && sudo reboot`

Option	Value/Example	Notes
astetcdir	`/etc/asterisk`	The location where the Asterisk configuration files are stored.
astmoddir	`/usr/lib/asterisk/modules`	The location where loadable modules are stored.
astvarlibdir	`/var/lib/asterisk`	The base location for variable state information used by various parts of Asterisk. This includes items that are written out by Asterisk at runtime.
astdbdir	`/var/lib/asterisk`	Asterisk will store its internal database in this directory as a file called `astdb`.
astkeydir	`/var/lib/asterisk`	Asterisk will use a subdirectory called `keys` in this directory as the default location for loading keys for encryption.
astdatadir	`/var/lib/asterisk`	This is the base directory for system-provided data, such as the sound files that come with Asterisk.
astagidir	`/var/lib/asterisk/` `agi-bin`	Asterisk will use a subdirectory called `agi-bin` in this directory as the default location for loading AGI scripts.
astspooldir	`/var/spool/asterisk`	The Asterisk spool directory, where voicemail, call recordings, and the call origination spool are stored.
astrundir	`/var/run/asterisk`	The location where Asterisk will write out its UNIX control socket as well as its process ID (PID) file.
astlogdir	`/var/log/asterisk`	The directory where Asterisk will store its log files.

Option	Value/Example	Notes
verbose	`3`	Sets the default verbose setting for the Asterisk logger. This value is also set by the -v command-line option. The verbose level is `0` by default.
debug	`3`	Sets the default debug setting for the Asterisk logger. This value is also set by the -d command-line option. The debug level is `0` by default.
alwaysfork	`yes`	Forking forces Asterisk to always run in the background. This option is set to `no` by default.
nofork	`yes`	Forces Asterisk to always run in the foreground. This option is set to `no` by default.
quiet	`yes`	Quiet mode reduces the amount of output seen at the console when Asterisk is run in the foreground. This option is set to `no` by default.
timestamp	`yes`	Adds timestamps to all output except output from a CLI command. This option is set to `no` by default.
execincludes	`yes`	Enables the use of `#exec` in Asterisk configuration files. This option is set to `no` by default.
console	`yes`	Runs Asterisk in console mode. Asterisk will run in the foreground and will present a prompt for CLI commands. This option is set to `no` by default.
highpriority	`yes`	Runs the Asterisk application with real-time priority. This option is set to `no` by default.
initcrypto	`yes`	Loads keys from the `astkeydir` at startup. This option is set to `no` by default.^[a]
nocolor	`yes`	Suppresses color output from the Asterisk console. This is useful when saving console output to a file. This option is set to `no` by default.
dontwarn	`yes`	Disables a few warning messages. This option was put in place to silence warnings that are generally correct, but may be considered to be so obvious that they become an annoyance. This option is set to `no` by default.
dumpcore	`yes`	Tells Asterisk to generate a core dump in the case of a crash. This option is set to `no` by default.^[b]
languageprefix	`yes`	Configures how the prompt language is used in building the path for a sound file. By default, this is `yes`, which places the language before any subdirectories, such as `en/digits/1.gsm`. Setting this option to `no` causes Asterisk to behave as it did in previous versions, placing the language as the last directory in the path, (e.g. `digits/en/1.gsm`).
internal_timing	`yes`	Uses a timing source to synchronize audio that will be sent out to a channel in cases such as file playback or music on hold. This option is set to `yes` by default and should be left that way; its usefulness has greatly diminished over the last few major versions of Asterisk.
systemname	`my_` `system` `_name`	Gives this instance of Asterisk a unique name. When this has been set, the system name will be used as part of the `uniqueid` field for channels. This is incredibly useful if more than one system will be logging CDRs to the same database table. By default, this option is not set.
autosystemname	`yes`	Automatically sets the system name by using the hostname of the system. This option is set to `no` by default.
maxcalls	`100`	Sets a maximum number of simultaneous inbound channels. No limit is set by default.
maxload	`0.9`	Sets a maximum load average. If the load average is at or above this threshold, Asterisk will not accept new calls. No threshold is set by default.
maxfiles	`1000`	Set the maximum number of file descriptors that Asterisk is allowed to have open. The default limit imposed by the system is commonly 1024, which is not enough for heavily loaded systems. It is common to set this limit to a very high number. The default system-imposed limit is used by default.
minmemfree	`1`	Sets the minimum number of megabytes of free memory required for Asterisk to continue accepting calls. If Asterisk detects that there is less free memory available than this threshold, new calls will not be accepted. This option is not set by default.
cache_record_files	`yes`	When doing recording, stores the file in the `record_cache_dir` until recording is complete. Once complete, it will be moved into the originally specified destination. The default for this option is `no`.
record_cache_dir	`/tmp`	Sets the directory to be used when `cache_record_files` is set to `yes`. The default location is a directory called `tmp` within the `astspooldir`.
transmit_silence	`yes`	Transmits silence to the caller in cases where there is no other audio source. This includes call recording and the `Wait()` family of dialplan applications, among other things. The default for this option is `no`.^[c]
transcode_via_sln	`yes`	When building a codec translation path, forces signed linear to be one of the steps in the path. The default for this option is `yes`.
runuser	`asterisk`	Sets the system user that the Asterisk application should run as. This option is not set by default, meaning that the application will continue to run as the user that executed the application.
rungroup	`asterisk`	Sets the system group that the Asterisk application should run as. This option is not set by default.
lightbackground	`yes`	When using colors in the Asterisk console, it will output colors that are compatible with a light-colored background. This option is set to `no` by default, in which case Asterisk uses colors that look best on a black background.
documentation_language	`en_US`	The built-in documentation for Asterisk applications, functions, and other things is included in an external XML document. This option specifies the preferred language for documentation. If it is not available, the default of `en_US` will be used.
hideconnect	`yes`	Setting this option to `yes` causes Asterisk to not display notifications of remote console connections and disconnections at the Asterisk CLI. This is useful on systems where there are scripts that use remote consoles heavily. The default setting is `no`.
lockconfdir	`no`	When this option is enabled, the Asterisk configuration directory will be protected with a lock. This helps protect against having more than one application attempting to write to the same file at the same time. The default value is `no`.
^[a]If any of the keys require a passphrase, this will block the startup process of Asterisk. An alternative is to run keys init at the Asterisk command line. ^[b]This is critical for debugging crashes. However, Asterisk must be compiled with the `DONT_OPTIMIZE` option enabled in menuselect for the core dump to be useful. ^[c]There is an important caveat to note when this option is enabled. The silence is generated in uncompressed signed linear format, which means that it will have to be transcoded into the format that the caller’s channel expects. The result may be that transcoding is required for a call that would not normally require it.

Option	Value/Example	Notes
astctlpermissions	`0660`	Sets the permissions for the Asterisk control socket.
astctlowner	`root`	Sets the owner for the Asterisk control socket.
astctlgroup	`apache`	Sets the group for the Asterisk control socket.
astctl	`asterisk.ctl`	Sets the filename for the Asterisk control socket. The default is `asterisk.ctl`.

udpbindaddr value	Description
`192.168.100.50`	Bind to a specific IPv4 address.
`2001:db8::1`	Bind to a specific IPv6 address
`0.0.0.0`	Bind to all IPv4 addresses on the system.
`::`	Bind to all IPv4 and IPv6 addresses.

Option	Value/Example	Notes
pbx_realtime	`1.6`	In versions earlier than Asterisk 1.6.x, the `pbx_realtime` module would automatically convert pipe characters into commas for arguments to Asterisk applications. This is no longer done by default. To enable this previous behavior, set this option to `1.4`.
res_agi	`1.6`	In versions earlier than Asterisk 1.6.x, the `EXEC` AGI command would automatically convert pipe characters into commas for arguments to Asterisk applications. This is no longer done by default. To enable this previous behavior, set this option to `1.4`.
app_set	`1.6`	Starting with the Asterisk 1.6.x releases, the `Set()` application only allows setting the value of a single variable. Previously, `Set()` would allow setting more than one variable by separating them with a `&`. This was done to allow any characters in the value of a variable, including the `&` character, which was previously used as a separator. `MSet()` is a new application that behaves like `Set()` used to. However, setting this option to `1.4` makes `Set()` behave like `MSet()`.

Option	Value/Example	Notes
autoload	`yes`	Instead of explicitly listing which modules to load, you can use this directive to tell Asterisk to load all modules that it finds in the modules directory, with the exception of modules listed as not to be loaded using the `noload` directive. The default, and our recommendation, is to set this option to `yes`.
preload	`res_odbc.so`	Indicates that a module should be loaded at the beginning of the module load order. This directive is much less relevant than it used to be; modules now have a load priority built into them that solves the problems that this directive was previously used to solve.
load	`chan_sip.so`	Defines a module that should be loaded. This directive is only relevant if `autoload` is set to `no`.
noload	`chan_alsa.so`	Defines a module that should not be loaded. This directive is only relevant if `autoload` is set to `yes`.
require	`chan_sip.so`	Does the same thing as `load`; additionally, Asterisk will exit if this module fails to load for some reason.
preload-require	`res_odbc.so`	Does the same thing as `preload`; additionally, Asterisk will exit if this module fails to load for some reason.

Option	Value/Example	Notes
`format`	`wav49\|gsm\|wav`	For each format listed, Asterisk will create a separate recording in that format whenever a message is left. The benefit of this is that some transcoding steps may be saved if the stored format is the same as the codec used on the channel. We like wav because it is the highest quality, and wav49 because it is nicely compressed and easy to email. We don’t like gsm due to it’s scratchy sound, but it enjoys some popularity.^[a]
`serveremail`	`user@domain`	When an email is sent from Asterisk, this is the email address that it will appear to come from.^[b]
`attach`	`yes,no`	If an email address is specified for a mailbox, this determines whether the messages is attached to the email (if not, a simple message notification is sent).
`maxmsg`	`9999`	By default Asterisk will only allow a maximum of 100 messages to be stored per user. For users who delete messages, this is no problem. For people who like to save their messages, this space can get eaten up quickly. With the size of hard drives these days, you could easily store thousands of messages for each user, so our current thinking is to set this to the maximum and let the users manage things from there.
`maxsecs`	`0`	This type of setting was useful back in the days when a large voicemail system might have only 40 MB^[c] of storage: it was necessary to limit the system because it was easy to fill up the hard drive. This setting can be annoying to callers (although it does force them to get to the point, so some people like it). Nowadays, with terabyte drives becoming common, there is no reason not to set this to a high value. Two considerations are: 1) if a channel gets hung in a mailbox, it’s good to set some sort of value so it doesn’t stay there for days, but 2) if a user wants to use her mailbox to record notes to herself, she won’t appreciate it if you cut her off after three minutes. A setting somewhere between `600` seconds (10 minutes) and `3600` seconds (1 hour) will probably be about right.
`minsecs`	`4`	Many folks will hang up instead of leaving a message when they call somebody and get voicemail. Sometimes this hangup happens after recording has started, so the mailbox owner gets an annoying two-second message of somebody hanging up. This setting ensures that Asterisk will ignore messages that are shorter than the configured minimum length. You should take care not to set this to a value that is too high, though, because then a message like “Hey it’s me give me a call” (which can be said in less than one second) will get lost, and you’ll get complaints of messages disappearing. Three seconds seems to be about right. To discourage people from leaving ultra-short messages that might be discarded, you can request callers to identify themselves and leave some information about why they called.
`maxgreet`	`1800`	You can define the maximum greeting length if you want. Again, since storage is not a problem and setting this too low will annoy your more verbose users, we suggest setting this to a high value and letting your users figure it out an appropriate length for themselves.
`skipms`	`3000`	When listening to messages, users can skip ahead or backwards by pressing (by default) `*` and `#`. This setting indicates the length of the jump (in milliseconds).
`maxsilence`	`5`	This setting defines the maximum time for which the caller can remain silent before the recording is stopped. We like to set this setting to one second longer than `minsecs` (if you set it equal to or greater than `minsecs`, you will get a warning).
`silencethreshold`	`128`	You can fine-tune the silence sensitivity of Asterisk to better define what qualifies as silence. In practice, this is seldom a good idea, since you cannot control the volumes of all the calls you’ll be getting from different places. It’s best to leave this at the default.
`maxlogins`	`3`	This little security feature is intended to make brute-force attacks on your mailbox passwords more time-consuming. If a bad password is received this many times, voicemail will hang up and you’ll have to call back in to try again. Note that this will not lock up the mailbox. Patient snoopers can continue to try to log into your mailbox as many times as they like, they’ll just have to call back every third attempt. If you have a lot of ham-fingered users, you can set this to something like `5`.
`moveheard`	`yes`	This setting will move listened-to messages to the `Old` folder. We recommend leaving this at the default.
`forward` `_` `urgent` `_` `auto`	`no`	Setting this to `yes` will preserve the original urgency setting of any messages the user receives and then forwards on. If you leave it at `no`, users can set the urgency level themselves on messages that they forward.
`userscontext`	`default`	If you use the `users.conf` file (we don’t), you can define here the context where entries are registered.
`externnotify`	`/path/to/script`	If you wish to run an external app whenever a message is left, you can define it here.
`smdienable`	`no`	If you are using Asterisk as a voicemail server on a PBX that supports SMDI, you can enable it here.
`smdiport`	`/dev/ttyS0`	Here is where you would define the SMDI port that messages between Asterisk and the external PBX would pass across.
`externpass`	`/path/to/script`	Any time the password on a mailbox is changed, the script you define here will be notified of the `context`, `mailbox`, and `new password`. The script will then be responsible for updating `voicemail.conf` (the Asterisk voicemail app will not update the password if this parameter is defined).
`externpassnotify`	`/path/to/script`	Any time the password on a mailbox is changed, the script you define here will be notified of the `context`, `mailbox`, and `new password`. Asterisk will handle updating the password in `voicemail.conf`. If you have defined `externpass`, this option will be ignored.
`externpasscheck`	`/usr/local/bin/voicemailpwcheck.py`	See the sidebar following this table for a description of this option.
`directoryintro`	`dir-intro`	The `Directory()` dialplan application uses the `voicemail.conf` file to search by name from an auto attendant. There is a default prompt that plays, called `dir-intro`. If you want, you can specify a different file to play instead.
`charset`	`ISO-8859-1`	If you need a character set other than ISO-8859-1 (a.k.a Latin 1) to be supported, you can specify it here.
`adsifdn`	`0000000F`	Use this option to configure the Feature Descriptor Number.^[d]
`adsisec`	`9BDBF7AC`	Use this option to configure the security lock code.
`adsiver`	`1`	This specifies the ADSI voicemail application version number.
`pbxskip`	`yes`	If you do not want emails from your voicemail to have the string [PBX] added to the subject, you can set this to `yes`.
`fromstring`	`The Asterisk PBX`	You can use this setting to configure the `From:` name that will appear in emails from your PBX.
`usedirectory`	`yes`	This option allows users composing messages from their mailboxes to take advantage of the Directory.
`odbcstorage`	`<item from res_odbc.conf>`	If you want to store voice messages in a database, you can do that using the Asterisk `res_odbc` connector. Here, you would set the name of the item in the `res_odbc` file. For details, see Chapter 22, Clustering.
`odbctable`	`<table name>`	This setting specifies the table name in the database that the `odbcstorage` setting refers to. For details, see Chapter 22, Clustering.
`emailsubject`	`[PBX]: New` `message` `${VM_MSGNUM} in mailbox ${VM_MAILBOX}`	When Asterisk sends an email, you can use this setting to define what the `Subject:` line of the email will look like. See the `voicemail.conf.sample` file for more details.
`emailbody`	`Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX u might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n`	When Asterisk sends an email, you can use this setting to define what the body of the email will look like. See the `voicemail.conf.sample` file for more details.
`pagerfromstring`	`The Asterisk PBX`	We don’t actually know anybody who uses pagers anymore (nor can we recall having seen one in many years), but if you have one of these historical oddities and you want to customize what Asterisk sends with its pager notification, presumably you can do that with this. A very practical usage of this feature for short message voicemail notifications is to send a message to an email to SMS gateway.
`pagersubject`	`New VM`	As above.
`pagerbody`	`New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}`	The formatting for this uses the same rules as `emailbody`.
`emaildateformat`	`%A, %d %B %Y at %H:%M:%S`	This option allows you to specify the date format in emails. Uses the same rules as the C function `STRFTIME`.
`pagerdateformat`	`%A, %d %B %Y at %H:%M:%S`	This option allows you to specify the date format in pager. Uses the same rules as the C function `STRFTIME`.
`mailcmd`	`/usr/sbin/sendmail -t`	If you want to override the default operating system application for sending mail, you can specify it here.
`pollmailboxes`	`no, yes`	If the contents of mailboxes are changed by anything other than `app_voicemail` (such as external applications or another Asterisk system), setting this to `yes` will cause `app_voicemail` to poll all the mailboxes for changes, which will trigger proper message waiting indication (MWI) updates.
`pollfreq`	`30`	Used in concert with `pollmailboxes`, this option specifies the number of seconds to wait between mailbox polls.
`imapgreetings`	`no, yes`	This enables/disables remote storage of greetings in the IMAP folder. For more details, see Chapter 18, External Services.
`greetingsfolder`	`INBOX`	If you’ve enabled `imapgreetings`, this parameter allows you to define the folder your greetings will be stored in (defaults to INBOX).
`imapparentfolder`	`INBOX`	IMAP servers can handle parent folders in different ways. This field allows you to specify the parent folder for your mailboxes. For more details, see Chapter 7, Outside Connectivity.
^[a]The separator that is used for each format option must be the pipe (`\|`) character. ^[b]Sending email from Asterisk can require some careful configuration, because many spam filters will find Asterisk messages suspicious and will simply ignore them. We talk more about how to set email for Asterisk in Chapter 18, External Services. ^[c]Yes, you read that correctly: megabytes. ^[d]The Analog Display Services Interface is a standard that allows for more complex feature interactions through the use of the phone display and menus. With the advent of VoIP telephones, ADSI’s popularity has decreased in recent years.

Option	Value/Example	Notes
`tz`	`eastern`, `european`, etc.	Specifies the `zonemessages` name, as defined in the `[zonemessages]` section, discussed in the next section.
`locale`	`de_DE.utf8`, `es_US.utf8`, etc.	Used to define how Asterisk generates date/time strings in different locales. To determine the locales that are valid on your Linux system, type locale -a at the shell.
`attach`	`yes`, `no`	If an email address is specified for a mailbox, this determines whether the messages are attached to the email notifications (otherwise, a simple message notification is sent).
`attachfmt`	`wav49`, `wav`, etc.	If `attach` is enabled and messages are stored in different formats, this defines which format is sent with the email notifications. Often `wav49` is a good choice, as it uses a better compression algorithm and thus will use less bandwidth.
`saycid`	`yes`, `no`	This command will state the caller ID of the person who left the message.
`cidinternalcontexts`	`<context>`, `<another` `context>`	Any dialplan contexts listed here will be searched in an attempt to locate the mailbox context, so that the name associated with the mailbox number can be spoken. The voicemail box number needs to match the extension number that the call came from, and the voicemail context needs to match the dialplan context.^[a]
`sayduration`	`yes`, `no`	This command will state the length of the message.
`saydurationm`	`2`	Use this to specify the minimum duration of a message to qualify for its length being played back. For example, if you set this to `2`, any message less than 2 minutes in length will not have its length stated.
`dialout`	`<context>`	If allowed, users can dial out from their mailboxes. This is considered a very dangerous feature in a phone system (mainly because many voicemail users like to use `1234` as their password), and is therefore not recommended. If you insist on allowing this, make sure you have a second level of password in the dialplan where another password is specified. Even so, this is not a safe practice.
`sendvoicemail`	`yes`, `no`	This allows users to compose messages to other users from within their mailboxes.
`searchcontexts`	`yes`, `no`	This allows voicemail applications in the dialplan to not have to specify the voicemail context, since all contexts will be searched. This is not recommended.
`callback`	`<context>`	This specifies which dialplan context to use to call back to the sender of a message. The specified context will need to be able to handle dialing of numbers in the format in which they are received (for example, the country code may not be received with the caller ID, but might be required for the outgoing call).
`exitcontext`	`<context>`	There are options that allow the callers to exit the voicemail system when they are in the process of leaving a message (for example, pressing `0` to get an operator). By default, the context the caller came from will be used as the exit context. If desired, this setting will define a different context for callers exiting the voicemail system.
`review`	`yes`, `no`	This should almost always be set to `yes` (even though it defaults to `no`). People get upset if your voicemail system does not allow them to review their messages prior to delivering them.
`operator`	`yes`, `no`	Best practice dictates that you should allow your callers to “zero out” from a mailbox, should they not wish to leave a message. Note that an `o` extension (not “zero,” “oh”) is required in the `exitcontext` in order to handle these calls.
`envelope`	`no`, `yes`	You can have voicemail play back the details of the message before it plays the actual message. Since this information can also be accessed by pressing `5`, we generally set this to `no`.
`delete`	`no`, `yes`	After an email message notification is sent (which could include the message itself), the message will be deleted. This option is risky, because even though a message was emailed, it is no guarantee that it was received (spam filters seem to love to delete Asterisk voicemail messages). Point being: on a new system, leave this at `no` until you are certain that no messages are being lost due to spam filters.
`volgain`	`0.0`	This setting allows you to increase the volume of received messages. Volume used to be a problem in older releases of Asterisk, but has not been an issue for many years. We recommend leaving this at the default. The `sox` utility is required for this to work.
`nextaftercmd`	`yes`, `no`	This handy little setting will save you some time, as it takes you directly to the next message once you’ve finished dealing with the current message.
`forcename`	`yes`, `no`	This strange little setting will check if the mailbox password is the same as the mailbox number. If it is, it will force the user to change his voicemail password and record his name.
`forcegreetings`	`yes`, `no`	As above, but for greetings.
`hidefromdir`	`no`, `yes`	If you wish, you can hide specific mailboxes from the `Directory()` application using this setting.
`tempgreetwarn`	`yes`, `no`	Setting this to `yes` will warn the mailbox owner that she has a temporary greeting set. This can be a useful reminder when people return from trips or vacations.
`passwordlocation`	`spooldir`	If you want, you can have mailbox passwords stored in the spool folder for each mailbox.^[b]
`messagewrap`	`no`, `yes`	If this is set to `yes`, when the user has listened to the last message, pressing next (`6`) will take him to the first message. Also, pressing previous (`4`) when at the first message will take the user to the last message.
`minpassword`	`6`	This option enforces a minimum password length. Note that this does not prevent the users from setting their passwords to something that’s easy to guess (such as `123456`).
`vm-password`	`custom_sound`	If you want, you can specify a custom sound here to use for the password prompt in voicemail.
`vm-newpassword`	`custom_sound`	If you want, you can specify a custom sound here to use for the “Please enter your new password followed by the pound key” prompt in voicemail.
`vm-passchanged`	`custom_sound`	If you want, you can specify a custom sound here to use for the “Your password has been changed” prompt in voicemail.
`vm-reenterpassword`	`custom_sound`	If you want, you can specify a custom sound here to use for the “Please reenter your password followed by the pound key” prompt in voicemail.
`vm-mismatch`	`custom_sound`	If you want, you can specify a custom sound here to use for the “The passwords you entered and reentered did not match” prompt in voicemail.
`vm-invalid-password`	`custom_sound`	If you want, you can specify a custom sound here to use for the “That is not a valid password. Please try again” prompt in voicemail.
`vm-pls-try-again`	`custom_sound`	If you want, you can specify a custom sound here to use for the “Please try again” prompt in voicemail.
`listen-control-forward-key`	`#`	You can use this setting to customize the fast forward key.
`listen-control-reverse-key`	`*`	You can use this setting to customize the rewind key.
`listen-control-pause-key`	`0`	You can use this setting to customize the pause/unpause key.
`listen-control-restart-key`	`2`	You can use this setting to customize the replay key.
`listen-control-stop-key`	`13456789`	You can use this setting to customize the interrupt playback key.
`backupdeleted`	`0`	This setting will allow you to specify how many deleted messages are automatically stored by the system. This is similar to a recycle bin. Setting this to `0` disables this feature. Up to 9999 messages can be stored, after which the oldest message will be erased each time another message is deleted.
^[a]Yes, we found this a bit confusing too. ^[b]Typically the spool folder is `/var/spool/asterisk/`, and it can be defined in `/etc/asterisk/asterisk.conf`.

Zone name	Value/Example	Notes
`eastern`	`America/New_York\|'vm-received' Q` `'digits/at' IMp`	This value would be suitable for the eastern time zone (EST/EDT).
`central`	`America/Chicago\|'vm-received' Q 'digits/at' IMp`	This value would be suitable for the central time zone (CST/CDT).
`central24`	`America/Chicago\|'vm-received' q 'digits/at' H N 'hours'`	This value would also be suitable for CST/CDT, but would play back the time in 24-hour format.
`military`	`Zulu\|'vm-received' q 'digits/at' H N 'hours'` `'phonetic/z_p'`	This value would be suitable for Universal Time Coordinated (Zulu time, formerly GMT).
`european`	`Europe/Copenhagen\|'vm-received' a d b 'digits/at' HM`	This value would be suitable for Central European time (CEST).

Argument	Purpose
`b`	Instructs Asterisk to play the busy greeting for the mailbox (if no busy greeting is found, the unavailable greeting will be played).
`d([c])`	Accepts digits to be processed by context `c`. If the context is not specified, it will default to the current context.
`g(#)`	Applies the specified amount of gain (in decibels) to the recording. Only works on DAHDI channels.
`s`	Suppresses playback of instructions to the callers after playing the greeting.
`u`	Instructs Asterisk to play the unavailable greeting for the mailbox (this is the default behavior).
`U`	Indicates that this message is to be marked as urgent. The most notable effect this has is when voicemail is stored on an IMAP server. In that case, the email will be marked as urgent. When the mailbox owner calls in to the Asterisk voicemail system, he should also be informed that the message is urgent.
`P`	Indicates that this message is to be marked as priority.

Argument	Purpose
`p`	Allows you to treat the <mailbox> parameter as a prefix to the mailbox number.
`g(#)`	Increases the gain by `#` decibels when playing back messages.
`s`	Skips the password check.
`a(folder)`	Starts the session in one of the following voicemail folders (defaults to `0`): `0 - INBOX, 1 - Old, 2 - Work, 3 - Family, 4 - Friends, 5 - Cust1, 6 - Cust2, 7 - Cust3, 8 - Cust4, 9 - Cust5`

What to change	Where to change it
Call progress tones	IP phones—on the phone itself ATAs—on the ATA itself Analog phones—DAHDI (`/etc/dahdi/system.conf`)
Type of PRI/BRI and protocol	DAHDI—`/etc/dahdi/system.conf` and `/etc/asterisk/chan_dahdi.conf`
Physical PSTN connections	Balun if required for PRI Get the analog pair to middle 2 pins of the RJ11 connecting to the Digium card
Caller ID on analog circuits	Asterisk—`/etc/asterisk/chan_dahdi.conf`
Prompt language and/or accent	Channel—`/etc/asterisk/sip.conf`, `/etc/asterisk/iax.conf`, `/etc/asterisk/chan_dahdi.conf`, etc. Dialplan—`CHANNEL(language)` function
Voicemail time/date stamps and pronunciation	Asterisk—`/etc/asterisk/voicemail.conf`
Tones delivered by Asterisk	Asterisk—`/etc/asterisk/indications.conf`

Time period (in seconds)	channel_1	channel_2	channel_3
0	`Dial(SIP/0000FFFF0001,20)`	`Wait(10)`	`Wait(15)`
5
10		`Dial(DAHDI/g0/14165551212)`
15			`Dial(SIP/MyITSP/12565551212,15)`
20	`Hangup()`
25
30			`Hangup()`
35
40

Option	Value/Example	Notes
`parkext`	`700`	Sets the default extension used to park calls.
`parkpos`	`701-720`	Sets the range of extensions used as the parking lot. Parked calls may be retrieved by dialing the numbers in this range.
`context`	`parkedcalls`	Sets the default dialplan context where the parking extension and the parking lot extensions are created.
`parkinghints`	`no`	Enables/disables automatic creation of dialplan hints for the parking lot extensions so that phones can subscribe to the state of extensions in the parking lot. The default is `no`.
`parkingtime`	`45`	Specifies the number of seconds a call will wait in the parking lot before timing out.
`comebacktoorigin`	`yes`	Configures the handling of timed-out parked calls. For more information on the behavior of this option, see the sidebar titled Handling Timed-Out Parked Calls with the comebacktoorigin Option.
`courtesytone`	`beep`	Specifies the sound file to be played to the parked caller when the parked call is retrieved from the parking lot.
`parkedplay`	`caller`	Indicates which side of the call to play the `courtesytone` to when a parked call is picked up. Valid options include `callee`, `caller`, `both`, or `no`. The default is `no`.
`parkedcalltransfers`	`caller`	Controls which side of a call has the ability to execute a DTMF-based transfer in the call that results from picking up a parked call. Valid options include `callee`, `caller`, `both`, or `no`. The default is `no`.
`parkedcallreparking`	`caller`	Controls which side of a call has the ability to execute a DTMF-based park in the call that results from picking up a parked call.^[a] Valid options include `callee`, `caller`, `both`, or `no`. The default is `no`.
`parkedcallhangup`	`caller`	Controls which side of a call has the ability to execute a DTMF-based hangup in the call that results from picking up a parked call. Valid options include `callee`, `caller`, `both`, or `no`. The default is `no`.
`parkedcallrecording`	`caller`	Controls which side of a call has the ability to initiate a DTMF-based one-touch recording in the call that results from picking up a parked call. Valid options include `callee`, `caller`, `both`, or `no`. The default is `no`.
`parkeddynamic`	`yes`	Enables the dynamic creation of parking lots in the dialplan. The channel variables `PARKINGDYNAMIC`, `PARKINGDYNCONTEXT`, and `PARKINGDYNPOS` need to be set.
`adsipark`	`yes`	Passes ADSI information regarding the parked call back to the originating set.
`findslot`	`next`	Configures the parking slot selection behavior. See Parking Lots for more details.
`parkedmusicclass`	`default`	Specifies the class to be used for the music on hold played to a parked caller. A music class set in the dialplan using the `CHANNEL(musicclass)` dialplan function will override this setting.
`transferdigittimeout`	`3`	Sets the number of seconds to wait for each digit from the caller executing a transfer.
`xfersound`	`beep`	Specifies the sound to be played to indicate that an attended transfer is complete.
`xferfailsound`	`beeperr`	Specifies the sound to be played to indicate that an attended transfer has failed to complete.
`pickupexten`	`*8`	Configures the extension used for call pickup.
`pickupsound`	`beep`	Specifies the sound to be played to indicate a successful call pickup attempt. No sound is played by default.
`pickupfailsound`	`beeperr`	Specifies the sound to be played to indicate a failed call pickup attempt. No sound is played by default.
`featuredigittimeout`	`1000`	Sets the number of milliseconds to wait in between digits pressed during a bridged call when matching against DTMF activated call features.
`atxfernoanswertimeout`	`15`	Configures the number of seconds to wait for the target of an attended transfer to answer before considering the attempt timed out.
`atxferdropcall`	`no`	Configures behavior of attended transfer call handling when the transferer hangs up before the transfer is complete and the transfer fails. By default, this option is set to `no` and a call will be originated to attempt to connect the transferee back to the caller that initiated the transfer. If set to `yes`, the call will be dropped after the transfer fails.
`atxferloopdelay`	`10`	Sets the number of seconds to wait in between callback retries if `atxferdropcall` is set to `no`.
`atxfercallbackretries`	`2`	Sets the number of callback attempts to make if `atxferdropcall` is set to `no`. By default, this is set to `2` callback attempts.
^[a]Read that again. It makes sense.

Asterisk™: The Definitive Guide

Leif Madsen

Jim Van Meggelen

Russell Bryant

Foreword

John Todd

Preface

Audience

Organization

Software

Conventions Used in This Book

Tip

Warning

Using Code Examples

Safari Books Online

Note

How to Contact Us

Acknowledgments

Leif Madsen

Jim Van Meggelen

Russell Bryant

Chapter 1. A Telephony Revolution

Asterisk and VoIP: Bridging the Gap Between Traditional and Network Telephony

The Zapata Telephony Project

Massive Change Requires Flexible Technology

Asterisk: The Hacker’s PBX

Asterisk: The Professional’s PBX

The Asterisk Community

The Asterisk Mailing Lists

Warning

Asterisk Wiki Sites

The IRC Channels

Asterisk User Groups

The Asterisk Documentation Project

The Business Case

Conclusion

Chapter 2. Asterisk Architecture

Modules

Note

Applications

Bridging Modules

Call Detail Recording Modules

Note

Channel Event Logging Modules

Channel Drivers

Codec Translators

Note

Format Interpreters

Dialplan Functions

PBX Modules

Resource Modules

Addon Modules

Test Modules

File Structure

Configuration Files

Modules

The Resource Library

The Spool

Logging

The Dialplan

Hardware

Asterisk Versioning

Previous Release Methodologies

The New Release Methodology

Conclusion

Chapter 3. Installing Asterisk

Installation Cheat Sheet

Note

Note

Note

Note

Distribution Installation

CentOS Server

Base system installation

Note

Base system update

Note

Enabling NTP for accurate system time

Adding a system user

Ubuntu Server

Option	Value/Example	Notes	Dial()/Queue() Flags
`blindxfer`	`#1`	Invokes a blind (unsupervised) transfer	`T, t`
`disconnect`	`*0`	Hangs up the call	`H, h`
`automon`	`*1`	Starts recording of the current call using the `Monitor()` application (pressing this key sequence a second time stops the recording)	`W, w`
`atxfer`	`*2`	Performs an automated transfer	`T, t`
`parkcall`	`#72`	Parks a call	`K, k`
`automixmon`	`*3`	Starts recording of the current call using the `MixMonitor()` application (pressing this key sequence again stops the recording)	`X, x`

Option	Description	Discussion
`d`	Enables full-duplex audio	Sometimes referred to as “talkback paging,” the use of this option implies that the equipment that receives the page has the ability to transmit audio back at the same time as it is receiving audio. Generally, you would not want to use this unless you had a specific need for it.
`i`	Ignores attempts to forward the call	You would normally want this option enabled.
`q`	Does not play beep to caller (quiet mode)	Normally you won’t use this, but if you have an external amplifier that provides its own tone, you may want to set this option.
`r`	Records the page into a file	If you intended on using the same page multiple times in the future, you could record the page and then use it again later by triggering it using `Originate()` or using the `A(x)` option to `Page()`.
`s`	Dials a channel only if the device state is `NOT_INUSE`	This option is likely only useful (and reliable) on SIP-bound channels, and even so may not work if a single line is allowed multiple calls on it. Therefore, don’t rely on this option in all cases.
`A(x)`	Plays announcement `x` to all participants	You could use a previously recorded file to be played over the paging system. If you combined this with `Originate()` and `Record()`, you could implement a delayed paging system.
`n`	Does not play announcement simultaneously to caller (implies `A(x)`)	By default the system will play the paged audio to both the caller and the callee. If this option is enabled, the paged audio will not be played to the caller (the person paging).

Name	Description	Example
Service	Symbolic name of service	`_sip.`
Proto	Transport protocol	`_udp.`
Name	Domain name for this record^[a]	`shifteight.org.`
TTL	Time to live (in seconds)	`86400`
Class	DNS class field (always `IN`)	`IN`
Priority	Target host priority	`0`
Weight	Relative weight of this record	`0`
Port	TCP/UDP port number	`5060`
Target	Hostname of machine providing this service	`pbx.shifteight.org.`
^[a]Note the trailing dot.

Name	Extension	Context
`leif`	`100`	`publicExtensions`
`leif.madsen`	`100`	`publicExtensions`
`lmadsen`	`100`	`publicExtensions`
`jim`	`101`	`publicExtensions`
`reception`	`0`	`Services` ^[a]
`voicemail`	`*98`	`Services`
^[a]Make sure this context exists on your system.

Options	Available values	Description
`persistentmembers`	`yes`, `no`	Set this to `yes` to store dynamically added members to queues in the AstDB so that they can be re-added upon Asterisk restart.
`autofill`	`yes`, `no`	With `autofill` disabled, the queue application will attempt to deliver calls to agents in a serial manner. This means only one call is attempted to be distributed to agents at a time. Additional callers are not distributed to agents until that caller is connected to an agent. With `autofill` enabled, callers are distributed to available agents simultaneously.
`monitor-type`	`MixMonitor`, <`unspecified`>	If you specify the value `MixMonitor` the `MixMonitor()` application will be used for recording calls within the queue. If you do not specify a value or comment the option out, the `Monitor()` application will be used instead.
`updatecdr`	`yes`, `no`	Set this to `yes` to populate the `dstchannel` field of the CDR records with the name of a dynamically added member on answer. The value is set with the `AddQueueMember()` application. This option is used to mimic the behavior of `chan_agent` channels.
`shared_lastcall`	`yes`, `no`	This value is used for members logged into more than one queue to have their last call be the same across all queues, in order for the queues to respect the wrap up time of other queues.

Options	Available values	Description
`musicclass`	Music class as defined by `musiconhold.conf`	Sets the music class to be used by a particular queue. You can also override this value with the `CHANNEL(musicclass)` channel variable.
`announce`	Filename of the announcement	Used for playing an announcement to the agent that answered the call, typically to let him know what queue the caller is coming from. Useful when the agent is in multiple queues, especially when set to auto-answer the queue.
`strategy`	`ringall`, `leastrecent`, `fewestcalls`, `random`, `rrmemory`, `linear`, `wrandom`	`ringall`: rings all available callers (default) `leastrecent`: rings the interface that least recently received a call `fewestcalls`: rings the interface that has completed the fewest calls in this queue `random`: rings a random interface `rrmemory`: rings members in a round-robin fashion, remembering where we left off last for the next caller `linear`: rings members in the order specified, always starting at the beginning of the list `wrandom`: rings a random member, but uses the members’ penalties as a weight.
`servicelevel`	Value in seconds	Used in statistics to determine the service level of the queue (calls answered within the service level time frame).
`context`	Dialplan context	Allows a caller to exit the queue by pressing a single DTMF digit. If a context is specified and the caller enters a number, that digit will attempt to be matched in the context specified, and dialplan execution will continue there.
`penaltymemberslimit`	Value of `0` or greater	Used to disregard penalty values if the number of members in the queue is lower than the value specified.
`timeout`	Value in seconds	Specifies the number of seconds to ring a member’s device. Also see `timeoutpriority`.
`retry`	Value in seconds	Specifies the number of seconds to wait before attempting the next member in the queue if the `timeout` value is exhausted while attempting to ring a member of the queue.
`timeoutpriority`	`app`, `conf`	Used to control the priority of the two possible `timeout` options specified for a queue. The `Queue()` application has a timeout value that can be specified to control the absolute time a caller can be in the queue. The `timeout` value in `queues.conf` controls the amount of time (along with `retry`) to ring a member for. Sometime these values conflict, so you can control which value takes precedence. The default is `app`, as this is the way it works in previous versions.
`weight`	Value of `0` or higher	Defines the weight of a queue. A queue with a higher weight defined will get first priority when members are associated with multiple queues.
`wrapuptime`	Value in seconds	The number of seconds to keep a member unavailable in a queue after completing a call.
`autofill`	`yes`, `no`	Same as defined in the `[general]` section. This value can be defined per queue.
`autopause`	`yes`, `no`, `all`	Enables/disables the automatic pausing of members who fail to answer a call. A value of `all` causes this member to be paused in all queues she is a member of.
`maxlen`	Value of `0` or higher	Specifies the maximum number of callers allowed to be waiting in a queue. A value of zero means an unlimited number of callers are allowed in the queue.
`setinterfacevar`	`yes`, `no`	If set to `yes`, the following channel variables will be set just prior to connecting the caller with the queue member: `MEMBERINTERFACE`: the member’s interface, such as `Agent/1234` `MEMBERNAME`: the name of the member `MEMBERCALLS`: the number of calls the interface has taken `MEMBERLASTCALL`: the last time the member took a call `MEMBERPENALTY`: the penalty value of the member `MEMBERDYNAMIC`: indicates whether the member was dynamically added to the queue or not `MEMBERREALTIME`: indicates whether the member is included from real time or not
`setqueueentryvar`	`yes`, `no`	If set to `yes`, the following channel variables will be set just prior to the call being bridged: `QEHOLDTIME`: the amount of time the caller was held in the queue `QEORIGINALPOS`: the position the caller originally entered the queue at
`setqueuevar`	`yes`, `no`	If set to `yes`, the following channel variables will be set just prior to the call being bridged: `QUEUENAME`: the name of the queue `QUEUEMAX`: the maximum number of calls allowed in this queue `QUEUESTRATEGY`: the strategy method defined for the queue `QUEUECALLS`: the number of calls currently in the queue `QUEUEHOLDTIME`: the current average hold time of callers in the queue `QUEUECOMPLETED`: the number of completed calls in this queue `QUEUEABANDONED`: the number of abandoned calls `QUEUESRVLEVEL`: the queue service level `QUEUESRVLEVELPERF`: the queue’s service level performance
`membermacro`	Name of a macro defined in the dialplan	Defines a macro to be executed just prior to bridging the caller and the queue member.
`announce-frequency`	Value in seconds	Defines how often we should announce the caller’s position and/or estimated hold time in the queue. Set this value to zero to disable.
`min-announce-frequency`	Value in seconds	Specifies the minimum amount of time that must pass before we announce the caller’s position in the queue again. This is used when the caller’s position may change frequently, to prevent the caller hearing multiple updates in a short period of time.
`periodic-announce-frequency`	Value in seconds	Indicates how often we should make periodic announcements to the caller.
`random-periodic-announce`	`yes`, `no`	If set to `yes`, will play the defined periodic announcements in a random order. See `periodic-announce`.
`relative-periodic-announce`	`yes`, `no`	If set to `yes`, the `periodic-announce-frequency` timer will start from when the end of the file being played back is reached, instead of from the beginning. Defaults to `no`.
`announce-holdtime`	`yes`, `no`, `once`	Defines whether the estimated hold time should be played along with the periodic announcements. Can be set to `yes`, `no`, or only `once`.
`announce-position`	`yes`, `no`, `limit`, `more`	Defines whether the caller’s position in the queue should be announced to her. If set to `no`, the position will never be announced. If set to `yes`, the caller’s position will always be announced. If the value is set to `limit`, the caller will hear her position in the queue only if it is within the limit defined by `announce-position-limit`. If the value is set to `more`, the caller will hear her position if it is beyond the number defined by `announce-position-limit`.
`announce-position-limit`	Number of zero or greater	Used if you’ve defined `announce-position` as either `limit` or `more`.
`announce-round-seconds`	Value in seconds	If this value is nonzero, we’ll announce the number of seconds as well, and round them to the value defined.
`queue-thankyou`	Filename of prompt to play	If not defined, will play the default value (“Thank you for your patience”). If set to an empty value, the prompt will not be played at all.
`queue-youarenext`	Filename of prompt to play	If not defined, will play the default value (“You are now first in line”). If set to an empty value, the prompt will not be played at all.
`queue-thereare`	Filename of prompt to play	If not defined, will play the default value (“There are”). If set to an empty value, the prompt will not be played at all.
`queue-callswaiting`	Filename of prompt to play	If not defined, will play the default value (“calls waiting”). If set to an empty value, the prompt will not be played at all.
`queue-holdtime`	Filename of prompt to play	If not defined, will play the default value (“The current estimated hold time is”). If set to an empty value, the prompt will not be played at all.
`queue-minutes`	Filename of prompt to play	If not defined, will play the default value (“minutes”). If set to an empty value, the prompt will not be played at all.
`queue-seconds`	Filename of prompt to play	If not defined, will play the default value (“seconds”). If set to an empty value, the prompt will not be played at all.
`queue-reporthold`	Filename of prompt to play	If not defined, will play the default value (“Hold time”). If set to an empty value, the prompt will not be played at all.
`periodic-announce`	A set of periodic announcements to be played, separated by commas	Prompts are played in the order they are defined. Defaults to `queue-periodic-announce` (“All representatives are currently busy assisting other callers. Please wait for the next available representative”).
`monitor-format`	`gsm`, `wav`, `wav49`, <`any valid file format`>	Specifies the file format to use when recording. If `monitor-format` is commented out, calls will not be recorded.
`monitor-type`	`MixMonitor`, <`unspecified`>	Same as `monitor-type` as defined in the `[general]` section, but on a per-queue basis.
`joinempty`	`paused`, `penalty`, `inuse`, `ringing`, `unavailable`, `invalid`, `unknown`, `wrapup`	Controls whether a caller is added to the queue when no members are available. Comma-separated options can be included to define how this option determines whether members are available. The definitions for the values are: `paused`: members are considered unavailable if they are paused. `penalty`: members are considered unavailable if their penalties are less than `QUEUE_MAX_PENALTY`. `inuse`: members are considered unavailable if their device `status` is In Use. `ringing`: members are considered unavailable if their device status is `Ringing`. `unavailable`: applies primarily to agent channels; if the agent is not logged in but is a member of the queue, it is considered unavailable. `invalid`: members are considered unavailable if their device status is `Invalid`. This is typically an error condition. `unknown`: members are considered unavailable if device status is unknown. `wrapup`: members are considered unavailable if they are currently in the wrapup time after the completion of a call.
`leavewhenempty`	`paused`, `penalty`, `inuse`, `ringing`, `unavailable`, `invalid`, `unknown`, `wrapup`	Used to control whether callers are kicked out of the queue when members are no longer available to take calls. See `joinempty` for more information on the assignable values.
`eventwhencalled`	`yes`, `no`, `vars`	If set to `yes`, the following manager events will be sent to the Asterisk Manager Interface (AMI): `AgentCalled` `AgentDump` `AgentConnect` `AgentComplete` If set to `vars`, all channel variables associated with the agent will also be sent to the AMI.
`eventmemberstatus`	`yes`, `no`	If set to `yes`, the `QueueMemberStatus` event will be sent to AMI. Note that this may generate a lot of manager events.
`reportholdtime`	`yes`, `no`	Enables reporting of the caller’s hold time to the queue member prior to bridging.
`ringinuse`	`yes`, `no`	Used to avoid sending calls to members whose status is `In Use`. Recall from our discussion in the preceding section that only the SIP channel driver is currently able to accurately report this status.
`memberdelay`	Value in seconds	Used if you want there to be a delay prior to the caller and queue member being connected to each other.
`timeoutrestart`	`yes`, `no`	If set to `yes`, resets the timeout for an agent to answer if either a `BUSY` or `CONGESTION` status is received from the channel. This can be useful if the agent is allowed to reject or cancel a call.
`defaultrule`	Rule as defined in `queuerules.conf`	Associates a queue rule as defined in `queuerules.conf` to this queue, which is used to dynamically change the minimum and maximum penalties, which are then used to select an available agent. See the section called “Changing Penalties Dynamically (queuerules.conf)”.
`member`	Device	Used to define static members in a queue. To define a static member, you supply its `Technology`/`Device_ID` (e.g., `Agent/1234`, `SIP/0000FFFF0001`, `DAHDI/g0/14165551212`).

Options	Available values	Description
`maxloginretries`	Integer value	Specifies the maximum number of tries an agent has to log in before the system considers it a failed attempt and ends the call. Defaults to 3.
`autologoff`	Value in seconds	Specifies the number of seconds for which an agent’s device should ring before the agent is automatically logged off.
`autologoffunavail`	`yes`, `no`	If set to `yes`, the agent is automatically logged off when the device being called returns a status of CHANUNAVAIL.
`ackcall`	`yes`, `no`	If set to `yes`, the agent must enter a single DTMF digit to accept the call. To be used in conjunction with `acceptdtmf`. Defaults to `no`.
`acceptdtmf`	Single DTMF character	Used in conjunction with `ackcall`, this option defines the DTMF character to be used to accept a call. Defaults to `#`.
`endcall`	`yes`, `no`	If set to `yes`, allows an agent to end a call with a single DTMF digit. To be used in conjunction with `enddtmf`. Defaults to `yes`.
`enddtmf`	Single DTMF character	Used in conjunction with `endcall`, this option defines the DTMF character to be used to end a call. Defaults to `*`.
`wrapuptime`	Value in milliseconds	Specifies the amount of time after disconnection of a caller from an agent for which the agent will not be available to accept another call. Used in situations where agents must perform a function after each call (such as entering call details into a log).
`musiconhold`	Music class as defined in `musiconhold.conf`	Defines the default music class agents listen to when logged in.
`goodbye`	Name of file (relative to `/var/lib/asterisk/sounds/<lang>/`)	Defines the default goodbye sound played to agents. Defaults to `vm-goodbye`.
`updatecdr`	`yes`, `no`	Used in call detail records to change the source channel field to the agent/`agent_id`.
`group`	Integer value	Allows you to define groups for sets of agents. The use of agent groups is essentially deprecated functionality that we do not recommend you use. If you define `group1`, you can use `Agent/@1` in `queues.conf` to call that group of agents. The call will be connected arbitrarily to one of those agents. If no agents are available, it will return back to the queue like any other unanswered call. If you use `Agent/:1`, it will wait for a member of the group to become available. The use of strategies has no effect on agent groups. Do not use these.
`recordagentcalls`	`yes`, `no`	Enables/disables the recording of agent calls. Disabled by default.
`recordformat`	File format (`gsm`, `wav`, etc.)	Defines the format to be used when recording agent calls. Default is `wav`.
`urlprefix`	String (URL)	Accepts a string as its argument. The string can be formed as a URL and is appended to the start of the text to be added to the name of the recording.
`savecallsin`	Filesystem path (e.g., `/var/calls/`)	Accepts a filesystem path as its argument. Allows you to override the default path of `/var/spool/asterisk/monitor/` with one of your choosing.^[a]
`custom_beep`	Name of file (relative to `/var/lib/asterisk/sounds/<lang>/`)	Accepts a filename as its argument. Can be used to define a custom notification tone to signal to an always-connected agent that there is an incoming call.
`agent`	Agent definition (see description)	Defines an agent for use by `Queue()` and `AgentLogin()`. These are agents that will log in and stay connected to the system, waiting for calls to be delivered by the `Queue()` dialplan application. Agents are defined like so: agent => `agent_id`,`agent_password`,`name` An example of a defined agent would be: agent => 1000,1234,Danielle Roberts
^[a]Since the storage of calls will require a large amount of hard drive space, you will want to define a strategy to handle storing and managing these recordings. This location should probably reside on a separate volume, one with very high performance characteristics.

Value	Description
`paused`	Members are considered unavailable if they are paused.
`penalty`	Members are considered unavailable if their penalties are less than `QUEUE_MAX_PENALTY`.
`inuse`	Members are considered unavailable if their device status is `In Use`.
`ringing`	Members are considered unavailable if their device status is `Ringing`.
`unavailable`	Applies primarily to agent channels; if the agent is not logged in but is a member of the queue it is considered unavailable.
`invalid`	Members are considered unavailable if their device status is `Invalid`. This is typically an error condition.
`unknown`	Members are considered unavailable if device status is unknown.
`wrapup`	Members are considered unavailable if they are currently in the wrapup time after the completion of a call.

Value	Mapping (joinempty)	Mapping (leavewhenempty)
`yes`	(empty)	penalty,paused,invalid
`no`	penalty,paused,invalid	(empty)
`strict`	penalty,paused,invalid,unavailable	penalty,paused,invalid,unavailable
`loose`	penalty,invalid	penalty,invalid

Event	Information provided
`ABANDON`	Written when a caller in a queue hangs up before his call is answered by an agent. Three parameters are provided for `ABANDON`: the position of the caller at hangup, the original position of the caller when entering the queue, and the amount of time the caller waited prior to hanging up.
`ADDMEMBER`	Written when a member is added to the queue. The bridged channel name will be populated with the name of the channel added to the queue.
`AGENTDUMP`	Indicates that the agent hung up on the caller while the queue announcement was being played, prior to them being bridged together.
`AGENTLOGIN`	Recorded when an agent logs in. The bridged channel field will contain something like `Agent/9994` if logging in with `chan_agent`, and the first parameter field will contain the channel logging in (e.g., `SIP/0000FFFF0001`).
`AGENTLOGOFF`	Logged when an agent logs off, along with a parameter indicating how long the agent was logged in for.
`COMPLETEAGENT`	Recorded when a call is bridged to an agent and the agent hangs up, along with parameters indicating the amount of time the caller was held in the queue, the length of the call with the agent, and the original position at which the caller entered the queue.
`COMPLETECALLER`	Same as `COMPLETEAGENT`, except the caller hung up and not the agent.
`CONFIGRELOAD`	Indicates that the queue configuration was reloaded (e.g., via module reload app_queue.so).
`CONNECT`	Written when the caller and the agent are bridged together. Three parameters are also written: the amount of time the caller waited in the queue, the unique ID of the queue member’s channel to which the caller was bridged, and the amount of time the queue member’s phone rang prior to being answered.
`ENTERQUEUE`	Written when a caller enters the queue. Two parameters are also written: the URL (if specified) and the caller ID of the caller.
`EXITEMPTY`	Written when the caller is removed from the queue due to a lack of agents available to answer the call (as specified by the `leavewhenempty` parameter). Three parameters are also written: the position of the caller in the queue, the original position at which the caller entered the queue, and the amount of time the caller was held in the queue.
`EXITWITHKEY`	Written when the caller exits the queue by pressing a single DTMF key on his phone to exit the queue and continue in the dialplan (as enabled by the `context` parameter in `queues.conf`). Four parameters are recorded: the key used to exit the queue, the position of the caller in the queue upon exit, the original position the caller entered the queue at, and the amount of time the caller was waiting in the queue.
`EXITWITHTIMEOUT`	Written when the caller is removed from the queue due to timeout (as specified by the `timeout` parameter to `Queue()`). Three parameters are also recorded: the position the caller was in when exiting the queue, the original position of the caller when entering the queue, and the amount of time the caller waited in the queue.
`PAUSE`	Written when a queue member is paused.
`PAUSEALL`	Written when all members of a queue are paused.
`UNPAUSE`	Written when a queue member is unpaused.
`UNPAUSEALL`	Written when all members of a queue are unpaused.
`PENALTY`	Written when a member’s penalty is modified. The penalty can be changed through several means, such as the `QUEUE_MEMBER_PENALTY()` function, through using Asterisk Manager Interface, or the Asterisk CLI commands.
`REMOVEMEMBER`	Written when a queue member is removed from the queue. The bridge channel field will contain the name of the member removed from the queue.
`RINGNOANSWER`	Logged when a queue member is rung for a period of time, and the timeout value for ringing the queue member is exceeded. A single parameter will also be written indicating the amount of time the member’s extension rang.
`TRANSFER`	Written when a caller is transferred to another extension. Additional parameters are also written, which include: the extension and context the caller was transferred to, the hold time of the caller in the queue, the amount of time the caller was speaking to a member of the queue, and the original position of the caller when he entered the queue.^[a]
`SYSCOMPAT`	Recorded if an agent attempts to answer a call, but the call cannot be set up due to incompatibilities in the media setup.
^[a]Please note that when the caller is transferred using SIP transfers (rather than the built-in transfers triggered by DTMF and configured in `features.conf`), the `TRANSFER` event may not be reliable.