Is there a way with Twilio to:
have a customer call in and be placed in a hold queue
Twilio would then dial our CSR's telephone numbers sequentially (or all at once)
Connect whichever CSR answered to the customer
I can get the system to answer the phone, place the caller into a queue, but then dialing the CSR and then connecting the CSR to the caller is where I am stuck.
I can get the system to answer, play a message, and then dial a sequence of CSR numbers, but that is it.
BTW: I am using classic ASP, not that it really matters.
Twilio evangelist here.
To make Twilio dial out to your CSR's you're going to need to use the REST API, specifically POSTing to the Calls endpoint. Check out the docs for making calls.
In order to know if the CSR did not answer and you need to call another, you can use the StatusCallback parameter. This lets you specify a URL that Twilio will request when the call ends, letting you know why it ended. If the CSR does not answer then you can find that out in the StatusCallback request by checking the Status parameter and start a call to another CSR.
You could also start a bunch of outbound calls to a bunch of CSR's at the same time. If you do this you would need to track all of the calls (I'd just store each calls CallSid somewhere) and then when the call starts use the <Gather> verb to have the CSR indicate they want to accept the customers call. You would then just hangup on all of the other in-progress outbound calls and <Dial> that CSR into the Queue which would connect them to the customer.
Hope that helps.
Related
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.
Can anyone clarify what is the purpose of using queue ?
What i understand is that a webhook is just a URL , you do a POST request to that URL and then do some stuff based on the body/data of the request. So why i need to queue the data and store it in a database then loop through the database again and perform the stuff.
The short answer is, you don't have to use a queue. A webhook is just an HTTP request (typically POST) notifying your application of some type of event. The reason you might want to consider a queue is because of typical issues you could run into.
One of these is because of response time back to the webhook requester (source). Many sources want a response (HTTP status 200) as quickly as possible so they can dequeue the request from their webhook system. If processing the webhook takes some time, a source will typically advise you to use a queue to defer the lengthier process asynchronous to the 200 response to the webhook.
Another possible reason could be for removing duplicate requests. There is no guarantee with webhooks that you will only receive a single request per event. A queue can be used to de-dupe these requests.
I would recommend you stick with a simple request handler if possible, then evolve a more sophisticated handler if you run into issues. Consider queues as a potential design approach if you run into issues like those above.
You need some way to prevent a conflict if the webhook is invoked multiple times very close together.
It doesn't necessarily have to be a queue, though. If the webhook performs database queries and updates, you can use a transaction to ensure that this is atomic for each invocation.
In this respect, it's little different from any other web utility. You should do something similar in scripts that process web forms.
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.
I'm using asterisk 1.4
when my agent talks with customer , we need a option of putting their call in a IVR
in which agent and customer can listen those ivr menu at a same time , eventually customer should send some DTMF for IVR menus
Could any one pls tell me in what way we can achieve this
Consider dropping both calling and called parties (customer & agent, respective to how you're operating) into a conference call, and then!
...Go and originate another call (with AMI or a call file) that's operating from a new context/extension/proirity and also drop this originated call into the conference.
I haven't tested this with DTMF input into a call prompt -- But, I have played automated messages to both calling and called party in a conference using this technique.
The caveat may be that both calling and called parties will have control of the prompt. Which may or not be relevant to your use case.
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.