I was wondering, how can one get the list of ongoing calls in Asterisk when using Asterisk-Java?
It is needed when e.g. restarting ManagerAPI and I would like to display all the calls, which are currently running on Asterisk.
Previously it was done by observing properties on Channels created in Asterisk (I think it was 'linkedChannel' property), but this is now (Asterisk 13) not possible - linkedChannel is not updated when call is created.
You should issue via AMI COMMAND action with "core show channels concise" and collect output.
Related
I have a Python service which connects to Asterisk via AMI and listens for events to detect when a call has begun.
This seems to work on most of the Asterisk servers I connect to. However, on a few of our servers we just don't see any of the AMI events (e.g. Newstate) when the call happens, though we do later see the Cdr event once the call has completed.
I've confirmed that this isn't specific to the library we're using to connect to AMI (py-Asterisk), because I see exactly the same thing when I connect manually, e.g.
$ openssl s_client -connect my-asterisk-server:5039
...
Asterisk Call Manager/2.10.3
Action: Login
Username: manager
Secret: ThisIsWhereITypedTheActualPassword
Response: Success
Message: Authentication accepted
Event: FullyBooted
Privilege: system,all
Status: Fully Booted
Action: Events
EventMask: on
Response: Success
Events: on
In the above block, I manually connected to AMI, logged in as the same administrator my Python code is using, and ensured that all events are turned on (though my asterisk config should already be displaying all events I care about by default).
After this point what I see on some of my Asterisk servers is a series of expected events like Newstate, followed by an eventual Cdr event. On other servers, I see only the Cdr event, with nothing proceeding it. This is completely consistened within each server, meaning a server either always sends all of the expected events or it never does.
I've checked the versions of asterisk, the manager.conf config file, the extensions.conf dialplan, the asterisk console in verbose mode (i.e. connecting via asterisk -vvvr to the running process), and just generally comparing my config files to my actually-working Asterisk servers.
I'm stumped as to what could be causing this, or even what to try next. If it matters, here's what my manager.conf looks like:
[general]
tlsenable=yes
tlsbindaddr=10.0.0.123:5039
tlscertfile=/etc/pki/asterisk/ami.crt
tlsprivatekey=/etc/pki/asterisk/ami.key
[admin]
secret=TheActualPasswordIsOnThisLine
read=system,call,log,verbose,command,agent,user,originate,cdr
write=system,call,log,verbose,command,agent,user,originate,cdr
EDIT: After some further digging, it seems that the only events which are showing up are Cdr events, so even things like peer registration events don't show up. I've also confirmed that all of my Asterisk 13.19.0-1 servers are exhibiting this behavior, and the only working servers are running much older versions of Asterisk.
The weird thing is that calls do come through successfully, so the problem is not that I'm missing some necessary module. (Or maybe I am? Is there some "make events show up in AMI" module that I need to ensure is loaded?)
FURTHER EDIT: I was able to turn on CEL events (Channel Event Logging), and those events show up, but this is a different set of events than the standard Newchannel/Newstate/etc I'm looking for. In theory I could rewrite a large portion of my service to use the CEL events, but ideally I'd just turn on the standard Newstate/Newchannel/Hangup events.
It turns out the issue is that I was missing
enabled=yes
in my manager.conf. Hopefully this will be helpful to someone in the future: even if you're able to connect to AMI, log into AMI, receive some events from AMI, and send commands to AMI and get responses back, it might not be enabled, and this will suppress most of the core events like Newchannel, Newstate, etc.
One clue was that running manager show settings in the asterisk console returned this:
Global Settings:
----------------
Manager (AMI) No
...
which is apparently how it indicates that it's not enabled.
I use Asterisk 16.5 and sip trunk.
If known sip channel can i find uniqueid of call?
Note: I want do it with Asterisk AMI actions and events.
Action: Command
Command: core show channels concise
give you a list of :
Channel:Context:Exten:Priority:Stats:Application:Data:CallerID:Accountcode:Amaflags:Duration:Bridged
You have to pick the correct channel with the "Channel" field and make a
Action: Command
Command: core show channel YOUR_CHANNEL
If you expect do events, correct way is use event. I.e collect NewChannel events and build map(for example, in memcached).
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_Newchannel
Running too many commands sometimes result problem with ami connection.
As an example of such code you can check FOP opensource or FOP2 project(no source now). https://www.fop2.com/about.php
I logged in using agc (agent) panel in vicidial and just to test manual dial, I just executed following MySql query fron cmd.
UPDATE vicidial_live_agents set external_dial='12122351880!!YES!NO!YES!!1478530720!!!!!!' where user='1001';
After this , i got a a call on my softphone (already configured).
I am unable to understand one thing, How can an explicit entry to the vicidial_live_agents table of mysql, make a call ?
You shouldn't be messing with the database for calls.
If you configured everything correctly, you should recieve a call as soon as you log-in which will send you to a conference, you should hear a voice confirming it (you are the only one in this conference), all future calls will be connected to that conference, you should keep that call at all times and not control anything in the softphone.
If you need to make a manual calls you can either set the "Manual Dial Override" option in the campaign to "ALLOW ALL", which will enable a link in the agent screen labeled "Manual Dial" or you can use the agent_api with the external_dial function.
What you have found is the raw method that the Agent API uses to implement a manual dial call. For more information on how the Agent API works, and for all of the options you can use with the "external_dial" function, take a look at the Agent API documentation:
http://vicidial.org/docs/AGENT_API.txt
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 database where one entry is structured like so:
number_to_call date file_to_play
Using asterisk, I need to do the following:
1. Check the database daily.
2. If date matches that of today's, then initiate call on number.
3. Once phone has been picked up, play file_to_play.
Some general pointers on how I even begin to do this would be great.
Most of the applications that I have written so far have worked on incoming calls. I have the following questions:
1. How do I write a "daemon" that will check the database? Asterisk should have both user and group privileges for it to execute properly. How do I do this?
2. Can I initiate an outgoing call from outside of the asterisk dialplan?
The calls are being made to a PSTN/mobile number.
You can Write any script Which can check DB on daily basis and once it maches the date range you can initiate a call using .call files.
Please study asterisk auto-dial out from voip-info.org - I think you can understand better then.
You can run your script for same user as asterisk runs there is also one more method to initiate call from linux which we can call Originate CLI command which can also refer to http://voip-info.org/.
In general it is not a great idea to write your own dialer, unless your volume is very very low. Where I work, we started rolling our own but at the end went with a commercial solution that handled most of the logic. There are a number of commercial and free solutions out there, so don't reinvent the wheel.
See my answer to https://stackoverflow.com/questions/11666755/outbound-dialer-using-asterisk/14589901#14589901 for why it is not a good idea to roll your own.