I am wondering if there is an easy way to hangup a channel created with Originate?
What I do is following:
Call in dialplan triggers Agi;
Agi starts AMI: Originate with Channel "SIP/201".
Now the extension rings. At answer I can bridge the channels. But, if the calling party leaves the call before the call is answered, I would like to stop the outgoing call. When I send AMI Hangup with Channel "SIP/201", It can't hangup originated call.
I can stop that call using 'hangup request channelname' using CLI
but how to hangup call using program.
please help me. how to hangup originated call
If calling party leave call,it will be auto-hanguped with CANCEL cause.
You also can use AMI action Hangup
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Hangup
BTW, very likly you just doing it wrong way.
Related
My goal
Pass an incoming call directly to Stasis and allow the app to decide whether to play a ringing or busy tone to the caller.
The problem
With my ARI app, if I omit the same => n,Ringing line from my dialplan, the Stasis app returns an error if the caller hangs up. I can have a call hang without any early media, until I pass a channel.play() command, via the ARI.
This solution has 2 issues:
The Stasis app receives a second StasisStart when the caller hangs up, returning a Channel not found error.
There is no command for channel.busy
Does anyone have any suggestions?
My only option that I can currently see is to ensure that all users have voicemail and a busy tone is never played. Not everyone wants / likes voicemail and it is not ethical to answer the call and play a busy tone, without the caller knowing that their call is connected.
Update
Using the following dialplan, I can get this to work in the desired way (plays busy to a user if they're not available), but with an error:
extensions.conf
[public]
exten => _.,1,NoOp()
same => n,Stasis(myStasisApp, ${SIP_HEADER(to)})
same => n,Busy(10)
same => n,Hangup()
myApp.js
// The user is available
channel.ring();
// The user is busy
channel.continueInDialplan();
Error
Another StasisStart is sent when the caller hangs up, followed by:
Unhandled rejection Error: {
"message": "Channel not found"
}
We faced the same problem and lost precious time finding out the reason so I'm sharing the solution here and maybe it will help.
extensions.conf
[public]
exten => _.,1,NoOp()
same => n,Stasis(myStasisApp)
same => n,Hangup()
When Asterisk receive a call, it will start the stasis app.
Create a Bridge.
Add the incomming channel A in this bridge.
Create a new outgoing channel B from your ari app with POST /channels/create.
Add the outgoing channel B in that bridge.
Dial from channel B the destination, where both of the channels
are in the same bridge with POST /channels/{channelId}/dial
Now, you will be able to see a new ARI Dial events with Ringing and Answer.
For the Hangup, you will receive Channels end events with Hangup Cause Code not a text like 17 for busy
Asterisk Hangup Cause Mappings
It's simpler to originate a channel (Asterisk version 13) instead of create and dial (Asterisk version 14) but you will not have the early media or a full control on that channel because it's created by Asterisk and not the ARI app so this channel will start sending event back to ARI when the call start and not before.
Asterisk 14 ARI: Create, Bridge, Dial.
ARI and Channels: Manipulating Channel State
This thread helped us a lot:
Re: ARI: add channel to bridge immediatelly after originate action
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Have fun ! and Hope this will help.
/ohammami
How can I execute a script to open our CRM app on the specific client CALLERID in when the call is answered in asterisk (on the computer of the receiver of the call and not the server asterisk) ?
I can execute a script on the server, but can't do it on the client that reveives the call.
Thanks a lot.
You probably need asterisk events
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AMI+Events
Event can be AgentConnect if you use queues or Join/Bridge if you need it fire without queues.
You need write always-running daemon which will fire your scripts on event.
I have a toll free DID that users call to access my PBX service on an Asterisk box. The problem is; this DID comes only with a single channel so the system can only receive one call at a time. My initial idea was to simply get the caller ID of the incoming call, disconnect the caller and issue an automated call back to him to proceed with the call. This would free up my toll free number but could be confusing for the caller of course and also, there are issues where the caller calls from behind an extension. The best solution would be to somehow seemlessly switch the call to an outgoing trunk to reconnect the caller but now using my SIP trunk.
My question is; is there a way to do this in Asterisk (or I guess, does SIP somehow allow such operation)?
Thanks in advance.
That is called "callback".
Yes, you can do it. No, asterisk have no internal way do that and no way do it not noticable for user.
I am using Adhearsion on top of Asterisk (version 11.9.0).
To make outbound calls Adhearsion uses AMI originate command. Problem is Asterisk doesn't say why the call got hung up.
If the callee is busy or did not pick up the call or hung up the call or switched off i am getting the same reason code ( 0 ).
Is there a way to get different Hung up reasons?
You can do originate to channel like Local/number#some_context/n
After that you can write some_context to dialout and handle in that context usual way any dial cause.
I'm adjusting simple application that among other things should be able to call another party using Asterisk AMI Originate command.
I'm stuck and I believe that my originate command is wrong.
Where/how can I see log of Originate commands that Asterisk creates when I use regular phone so I can compare it to my hand crafted one?
Use a network sniffer, such as tcpdump or wireshark, and capture the packets that come and go to/from asterisk. By default, it uses 5038/tcp. Check your manager.conf file, and look for the bindaddr and port options to be sure you capture the right traffic.
If you are using ssl (sslenable=yes), then you will have to configure wireshark with your ssl keys, so it can decrypt the traffic or just use normal tcp (without ssl) for debugging and then switch to ssl.
You should see the Action: Originate coming in to asterisk, and the asterisk response and the associated events. Look for the ActionID parameter of the action so you can trace which responses and events correspond to each issued action.
Take into account that an async originate (async: true) will return a response as soon as the action is received by asterisk, but it will then send asynchronous events to inform the call status (once finished). On the other hand, when using async: false, the call will be placed and the response will have the status.
A few more resources on the originate action:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Originate
Another question related to the async parameter:
Asterisk originate response says successfully queued but nothing more
Hope it helps!
EDIT: Asterisk does not create the originate command, but will dial a target (a channel) based on an incoming originate action, or call file, so your application (the ami client) will issue an originate action and then asterisk will react to it by doing the call. If your call is originating from a phone, it's more probable that the call is being originated by a dial() command in your dialplan.