I`m trying to build a script that will capture the credit card info like card number,cvc and expiration date using asterisk 11.x and asterisk-java library for AMI/AGI integration.
Right now I am able to build a script that will acquire that info if it is called via dialplan but i have a different scenario:
1. A call enters a queue.
2. An agent from the specific queue answer the call
3. The caller wants to input the card details
4. After the caller has entered the card details is redirected back to agent to continue the call.
My specific problem is related to step 3 as I do not know how to route the caller to my AGI and then back to the same agent. (eventually the agents has to be still involved in (some) call to guarantee that when the caller returns from agi it is still available)
Any idea how can I achieve that ? I know that this is a common practice so I think that there has to be a way.
When the call is delivered to the agent, use a macro to set a custom channel variable with the agent ID or extension in it.
Then, when your credit-card authentication function is done, read the variable and use an AGI command to transfer the call back to the agent.
Further Reading
http://www.voip-info.org/wiki/view/Asterisk+variables
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Transfer
Note if this solution solves your problem, please 'accept' it to make it easier for others with the same issue to find it. thanks!
There are no any common practice for business process like you have. That depend of you and your client only.
You can use features conf or transfer. Can transfer to special extension or to conference room.
No way say what suite you better.
For sure you need understand how asterisk work before write any AGI/AMI or dialplan application. I can recommend ORelly's "Asterisk the future of telephony" book as start point.
Related
I've got a basic twilio setup using studio for a simple IVR (like less than 20 minute setup kind of simple).
One thing I'm doing is using simulring to hit multiple parties and whoever answers. The issue I'm having is that if there is a voicemail to be hit, it appears to be ok with that and then plays the endpoints custom voicemail.
That's not really good with our setup.
Do you guys know how I can trigger the calls to multiple phones, detect a real person and then transfer the call, otherwise trigger a voicemail?
I can't imagine this is unique.
Twilio developer evangelist here.
There are two options you can use here, either call screening/human detection, in which you ask the person answering the phone to, for example, dial 1. A voicemail won't do this and you can leave a message or hang up, a human will dial the number and you can then take them on to the rest of the call. Or there is answering machine detection (AMD). When making an outbound call with AMD, once detected Twilio will make the webhook callback with a parameter that describes whether the call was answered by a human or a machine.
I recommend reading this article on both options, which should help you to understand what will work best for your use-case.
We receive international calls into an Asterisk server (13.20) where some of the calls are automated, meaning there is no person involved, sort of M2M.
It is important for us to know where those automated call are coming from. Since it is easy to generate a call with faked ID we want to strengthen the authentication by identifying the original network from where the call was made.
When looking at the Asterisk logs I can see that a call came from Twilio for example, but that's it, no more tracking information.
My question:
Is it possible to track a call backwards beyond the last PBX who transferred the call to my server?
Some operators send some tracking in sip headers
For see more info, check sip debug.
asterisk -r
sip set debug on
However most of operators not provide for clients info about path of calls, some even not store it for internal use.
I have 2 servers, one with QueueMetrics installed and the other with Asterisk and qLoader. Both of these servers are able to communicate with each other and pass data back and forth. The problem that I am running into is that when adding an agent to a queue, neither Queuemetrics or the Asterisk server is recognizing that the agent is being added. I can make a call into the queue and see it being offered to the queue in real-time in QM. When I try to add an agent to the queue, QM says that the agent is being added. When watching the real-time monitor after the agent is added, the agent just never shows up as logged in. I have checked the MySQL database as well and it never shows the agent as being logged in either. I am unsure at this point what is causing the issue, and whether it is that the agent isn't being add to MySQL because of QM or because the agent available isn't being sent to the Asterisk server.
Any help would be greatly appreciated.
i think you need read something about queues setup and check your setup. For example this book: http://cdn.oreilly.com/books/9780596510480.pdf
Unfortanly no way troubleshoot your install without see config&debug.
QueueMetrics uses two different channels to work with the server: it reads data from the queue_log through qloader (and that is working for you) and uses AMI plus a custom Asterisk dialplan to perform actions like logging-on agents.
You can test the AMI and whether the dialplan is included through the DbTester tool - see http://queuemetrics.com/manuals/QM_UserManual-chunked/ar01s20.html#DBTEST - note that you may need to edit the supplied dialplan to match the format of your channel names. Qm is very flexible, but you need to tell it how your environment is set-up.
Or just keep an Asterisk CLI open when trying to log in agents and see what happens.
I am new to Asterisk and Voip. I wanted to accomplish a following small thing using Asterisk.
Description
Asterisk is used as server
Several Voip clients. (Two types of clients possible. One which can start a conference call, other can't call but can only hear.) Only caller client can start/end this call.
The call can't be longer then a particular time.
Is it possible through Asterisk. How does asterisk help to implement this scenario. What does I need to learn? Any web links will be very helpful.
Thanks
You can do all that with Asterisk and ConfBridge:
http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge
Use the following options to accomplish your objectives:
'A' — Set marked mode
'm' — Set initially muted.
'w' — Wait until the marked user enters the conference
You can use another dial plan function: TIMEOUT(absolute) to limit the conference duration.
To start I would look at the examples in the above link.
I would like to know if there is a web api for asterisk. I would also like to know if the average wait time to talk to a customer service agent is exposed through the api.
I have looked around online, but could not get an firm answer.
Any pointers are appreciated.
AFAIK, no, there is no such thing in Asterisk.
What does exist is the ability of parsing the queue_log file. You can get the moment the call started, the moment the call was answered by an agent, and subtract them - this will give you the wait time. Also, the first extra data value of the CONNECT event contains the time waited.
(If you are not in the mood for parsing a text file, you can register the queue logs in the database and use SQL to generate reports based on the logs. This is in fact my preferred approach.)
If you want to provide this information to other apps, you can write your own application which reads queue_log file/table and provides a webservice which returns wait times. In the case you decide to do it, we can try some more robust answers.