I want to execute an AGI script for calling party after called party hangs up. For example, for doing a survey for customers I am going to run an AGI script after agent hangs up.
Unfortunately, when AGI scripts runs the agi debug output says:
"511: The command cannot be executed on a dead channel"
I use commands like "ANSWER" or "STREAM FILE" in my agi script which need a channel to run on.
I know that calling channel hangs up as soon as called party hangs up.
I tried DeadAGI instead of AGI and also "g" option in dial command but none of them works.
So, I think I have to search for a solution that keep the calling channel up so that I can run my script on that.
Any suggestion please?
You should not use "g" param for this case, because it is CALLER hangup. You should use "F"
F([[context^]exten^]priority): When the caller hangs up, transfer
the *called* party to the specified destination and *start* execution
at that location.
NOTE: Any channel variables you want the called channel to inherit
from the caller channel must be prefixed with one or two underbars ('_').
F: When the caller hangs up, transfer the *called* party to the next
priority of the current extension and *start* execution at that location.
NOTE: Any channel variables you want the called channel to inherit
from the caller channel must be prefixed with one or two underbars ('_').
NOTE: Using this option from a Macro() or GoSub() might not make
sense as there would be no return points.
g: Proceed with dialplan execution at the next priority in the current
extension if the destination channel hangs up.
Finally I found the answer. The problem is that we don't have to execute agi script in h extension otherwise we get an error for some functions like answer or streamfile in agi. we have to redirect the call to some extension before h extension and for that in elastix I had to change extensions.conf.
Related
I am using Asterisk 13.17.0 on a FreePBX 14.0.1.1 distro. I would like to execute a python script whenever a call gets picked up, regardless of whether it is an internal or external one, passing to it as command line parameters the number who's calling and the user who's picking it up. How could I modify my dialplan in order for this to be done? I've tried modifying the [app-pickup] extension in /etc/asterisk/extensions_additional.conf, but picking up an internal call with this modification was to no avail
You should check diall command params, especial G param
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
G( context^exten^priority ) - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one.
context
exten
priority
In freepbx you can change dial params in settings tab.
You also SHOULD NOT ever touch extensions_additional.conf, it even have warning on top of file. File will be rewrited every time you change config. You have do changes in extensions_custom.conf
I'm trying to find answer how to make Asterisk execute some command (my script) after confbridge's recording is finished
There is the next info in confbridge.conf:
record_conference=yes
Records the conference call starting when the first user enters the
room, and ending when the last user exits the room.
It records file well but I want it sending wav file via email.
Could anybody help me?
My config now looks like this (if it's necessary):
exten => 333,1,ConfBridge(100010,100010_bridge_profile,100010_user_profile)
Dialplan scripting is limited to events relating to each call channel. To get event info for other parts of asterisk (such as the ConfBridge application) you should hook into the Asterisk Manager Interface (AMI).
There are many libraries already created to make working with the AMI easier. (That site may be outdated. Refer to the official Asterisk Wiki whenever possible.)
The AMI event you're interested in is "ConfBridgeEnd". Docs here.
You can use h-extension after confbridge, in which you have check if confbridge still active(last user).
If yes, run your script via System call.
I have a calling application developed in PHP with AGI on Asterisk framework, below is the basic flow of application.
We receive a call from user at our Asterisk ss7 server and forward the same call to another user from our server.
I want to know the status of call forward to another user. Status means what happen with call between both users e.g. Hangup, Busy, Not Answered etc.
Not much detail on how you are placing the call or anything else but if you are performing a Dial() and want to know at the end of the call the result try ${DIALSTATUS}.
This will contain something like 'ANSWER', 'NOANSWER' or 'BUSY'. For the full list and more info check out http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
you can use CDR feature or use a diaplan to get the dialed status
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql
Is anybody knows , how we can Originate an external number call using PHP AGI script ?
You have got two possible options. One is use the "Originate" command. See http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Originate for an example.
The other one, which is what I favor in my solutions, since it does not require AMI, is using spooled call files. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files for how to do them. The trick, and I cannot stress it enough, is to create the spool file in /tmp and then "move" the file into the ../spool/asterisk/outgoing directory.
If you assign a unique channel variable key/value to the spooled call, you can then pick it up with your dial plan and route the call appropriately.
I want to pickup call in Asterisk using AMI. I can originate call, but totally don't know, how to answer the phone...
Script for calling:
#login
sock = socket.socket(af, socktype, proto)
sock.connect(sockaddr)
sock.send('Action: login\r\n')
sock.send('Events: off\r\n')
sock.send('Username: '+str(ast_server.login)+'\r\n')
sock.send('Secret: '+str(ast_server.password)+'\r\n\r\n')
#originate call
sock.send('Action: originate\r\n')
sock.send('Channel: ' + str(user.asterisk_chan_type) + '/' + str(user.internal_number)+'\r\n')
sock.send('Timeout: '+str(ast_server.wait_time*1000)+'\r\n')
sock.send('CallerId: '+str(user.callerid)+'\r\n')
sock.send('Exten: '+str(ast_number)+'\r\n')
sock.send('Context: '+str(ast_server.context)+'\r\n')
if ast_server.alert_info and user.asterisk_chan_type == 'SIP':
sock.send('Variable: SIPAddHeader=Alert-Info: '+str(ast_server.alert_info)+'\r\n')
sock.send('Priority: '+str(ast_server.extension_priority)+'\r\n\r\n')
#logout
sock.send('Action: Logoff\r\n\r\n')
time.sleep(1)
sock.close()
I need something similar, but for answering calls.
Can't find any useful command in *CLI> manager show command
Halp me, plox
You can't answer a call directly via AMI. This is because a new call will "arrive" at the given context/priority/extension configured in the dialplan (or it will be rejected if cant find one that applies). So whatever happens with that call will start at the given context/priority/extension in the dialplan.
If you want to handle calls via AMI, try using asynchronous AGI, like this:
exten => _X.,1,AGI(agi:async)
This will handle all calls to any extension that has at least 1 digit, by issuing an event (AsyncAGI) that you can handle with your AMI client.
Then, from your AMI client, you can send AGIAction's, like:
Action: AGI
Channel: SIP/adevice
Command: ANSWER
CommandID: MyCommandID
This will effectively allow you to run AGI commands (and handle a call like you would normally do in any AGI script) from your AMI client.
Hope it helps!