Hi,
Sangoma Netborder Call Analyzer.Asterisk support added in 2008.Works as a SIP Proxy Dialogic Diastar Perfect Call.Asterisk support added in 2010.Works as SIP Proxy and with chanwoomera Lumenvox Call Progress Analysis.New offering in 2012.Works as add-on to existing speech platform. I then use a DB look up in the dial plan to get the list of sound files. I have my DBGet Ruby script return two values; a 'more files' flag and a list of up to ten sound files. I loop the DBGet and keep playing sets of file to the customer as long as the 'more files' flag is true. It works like a charm. And keeps the call files -much- easier to. A call file is really simple, It just tells the PBX what channel to send to what extension, The below example would call a SIP registration named John and direct it to the extension in the context dialout. You create your call file and put it in the spool directory and asterisk will process it. Add the callfile information to a file such as 'callfile.new' or some other appropriately named file. Our dialplan will perform a lookup in the AstDB to determine which device to call, and will then call the device, and upon answer, Playback the silence/1 (1 second of silence) and the tt-weasels sound files. Rules and Examples. The asterisk is a punctuation mark that looks like a little star (. ). Click Here for Step-by-Step Rules, Stories and Exercises to Practice All English Tenses. The asterisk is made on your keyboard by holding the SHIFT key and pressing the 8 on the top number line. We use the asterisk in English writing to show that a footnote, reference or comment has been added to the.
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
New Phone
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).
The new phone in Phonelist
Note the username and the internal phone number (here 623).
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:MY!!Password@192.168.254.1/623
; 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.
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
If all is running as expected start asterisk as daemon
Michael
This article presents an improved script to move files (in this case telephone calls made through Asterisk) unturned open files (calls in progress). It is very typical to have to make recordings of calls from Asterisk in a Call Center for example.
To make this script I used as a base one that I found here, thanks to its author.
Remind before starting what colors mean Applications and Systems:
written texts in the command line: Blue
Output from the command line: Verde
File names and file content: brown
Justification
Asterisk is a telephony software (widely used in VoIP) for building a switchboard with multiple functionalities, including call recording. Normally, for performance, call recordings perform them on a RAM drive, or RAM disk (if you want to know how to create one you can follow this link), then, after certain time intervals, permanently stored on a disk drive, for instance, in a server NAS. Thus, also avoid the deterioration involved in the drive the continuous effort of writing by recording multiple calls.
Of course this is a risk; an electrical fault and recordings are lost. So should save from time to time those recordings in a definite place.
This script I've adapted to my needs. All recordings are saved in a subdirectory that is created every day, with the date thereof, format gsm. The script will initially looking for that subdirectory, he looks into files that are not being used, and finally moved to its final location.
If we did not care to look first if they are being used or not, se podrían mover archivos incompletos, con la pérdida de información que ello supone.
Desarrollo
El script que propongo y que a mí me funciona 😉 es el siguiente:
Call From Asterisk
You can download it here if you are more comfortable.
Asterisk Outgoing Call File Example
Do not forget to comment and contribute your customized solutions, you can always improve.