- Asterisk Call File Example Python
- Asterisk Call Manager
- Asterisk Call File Example Pdf
- Asterisk Call File Example Template
- Asterisk Call Status
In this short example, we test the StarAstAPI[48] in PHP, which assumes a PHP 5[49] that was compiled with
--enable-sockets
.[50] Unfortunately, the StarAstAPI files still contain the obsolete 'short open tags' (<?
). If you encounter them, replace them with the correct syntax (<?php
). Four demo scripts are included with the API: sLogin.php
attempts a login[51], sCommand.php
executes reload
on the CLI, sDial.php
tries a connection to SIP/120 and sEvents.php
receives events. If we connect to Asterisk using asterisk -vvvr
and simultaneously run php -q sLogin.php
to open a connection to the AMI[52], watching the CLI, we see: This failed because the user did not exist, yet the demo script still reports success:followed by the response packet:The StarAstAPI is, as you can see, not completely clean, but is simple enough that it can be improved easily. If we call php -q sEvents.php
- this time with the correct user - we see:As a test, we execute a reload
in the CLI, which is reflected in the PHP script output:Give your creativity free-reign! Write a small script that calls all your friends - in the middle of the night, of course!Hi,
asterisk is a complete VoIP/SIP solution but can also be used as a SIP client to send a prerecorded message. In this example I will use a Fritz!box as the upstream SIP Server for asterisk.
First create an IP Phone and an corrosponding User Account at the Fritz!box.
Open the Fritz!Box GUI, then open the Phonelist
First create an IP Phone and an corrosponding User Account at the Fritz!box.
Open the Fritz!Box GUI, then open the Phonelist
New Phone
Asterisk Call File Example Python
See more: how to make call from asterisk, asterisk dial out and play message, asterisk test call from cli, asterisk auto dial out, asterisk auto dialer open source, asterisk schedule call, asterisk call file multiple channels, asterisk call file examples, day counter list format drupal, asterisk ivr sound format, global format, format sending. In our example, the extension 100 will redirect an unanswered call to the voicemail after 5 seconds. In our example, the extension 200 does not have voicemail. In our example, if any extension dial 500 it will be sent to the Voicemail menu. Delete the content of the voicemail.conf configuration file. This is an extension module for the Asterisk Gateway Interface (AGI) that adds commands to allow the transfer of audio files to and from Asterisk via the AGI session. This is useful when using FastAGI from a remote host; sounds recorded by Asterisk may be retrieved by remote FastAGI-providing service, for example, or sound files required by the. Asterisk File Locations (debian). /etc/asterisk/ - Asterisk configuration files. /var/lib/asterisk/ - contains the astdb, firmware and keys. /usr/share/asterisk/sounds - in built asterisk sound prompts. /var/spool/asterisk/ - temporary files and voicemail files. /var/log/asterisk/ - Asterisk log files.
LAN IP phone and define a name
New SIP User
Deselect Phone numbers for which asterisk should not receive incoming calls (this would also work).
Asterisk Call Manager
The new phone in Phonelist
Note the username and the internal phone number (here 623).
Then install asterisk
Then install asterisk
Rename the default /etc/asterisk/sip.conf file an create a new one
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
language=de
pedantic=yes
allowsubscribe=yes
subscribecontext=default_1
notifyringing=yes
notifyhold=yes
limitonpeers=yes
useclientcode=no
sendcallinfo=yes
dtmfmode=auto
rtpkeepalive=5
permit=192.168.254.0/255.255.255.0
deny=0.0.0.0/0.0.0.0
alwaysauthreject=yes
allowguest=no
localnet=192.168.254.0/255.255.255.0
allow=alaw
sendrpid=yes
trustrpid=no
registertimeout=60
checkmwi=yes
nat=force_rport
register => myAsteriskPhone:[email protected]/623
; virt asterisk SIP phone for outgoing calls
[2000]
type=friend
context=asterisk-phones
secret=SIPPel!
host=dynamic
; For incoming calls, Sectionname must match the register directive
[myAsteriskPhone]
type=friend
context=sip-asterisk
defaultuser=myAsteriskPhone
fromuser=myAsteriskPhone
secret=MY!!Password
host=192.168.254.1
fromdomain=192.168.254.1
qualify=yes
Adjust the host and fromdomain variables with the IP Address of your Fritzbox. And change the localnet and permit properties to your subnet. At the register directive adjust the internal phone number (here 623) noted previously.
Then create a call file /root/mycall.call. Replace 012345678with the number you want to call.
Channel: SIP/012345678@myAsteriskPhone
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: asterisk-phones
Extension: 10
Callerid:2000
Move all default extension files
And create a /etc/asterisk/extensions.conf with sequences what to do if the mycall.call file is used. asterisk-phones is the context and 10 is the extension refered from the mycall.call file.
[others]
[asterisk-phones]
exten => 10,1,Answer()
exten => 10,n,Wait(2)
exten => 10,n,Playback(hello-world)
exten => 10,n,Wait(2)
exten => 10,n,Hangup()
For test purposes start asterisk in foreground. Ignore the load_modules warnings. Importend is the Asterisk Ready prompt.
Asterisk Call File Example Pdf
To test the setup, copy the mycall.call file to /var/spool/asterisk/outgoing
This should end up in a call. If the call would accepted a hello world greeting is spoken
You can also use your own speech files. It has to convert in the gsm specific format. ffmpeg creates a WAV and a PCM u-Law. Use one of them. I recommend an seperate folder under
Open /etc/asterisk/extensions.conf and adjust the Playback directive, omit the file extension!
[others]
[asterisk-phones]
exten => 10,1,Answer()
exten => 10,n,Wait(2)
exten => 10,n,Playback(/var/lib/asterisk/sounds/my/mySpeech)
exten => 10,n,Wait(2)
exten => 10,n,Hangup()
Note: When you change a config file you have to reload asterisk
Or
Asterisk Call File Example Template
If all is running as expected start asterisk as daemon
Asterisk Call Status
Michael