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.
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.
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.
Basicly, I'm trying to read a game's chat and catch actions from the user.
Here is the image which I will explain the situation with :
1: I took a message in the chat
2: I tryed to find it in the game's memory with Cheat-Engine
3: By examining every addresses where it was found, I ended up to this one, which contains the chat formated with what seems to be html..
That part is only the bottom part of the chat. (I see the rest of it if I scroll up)
So, I asked myself how could I read game variable to interact with the game.
Another thing I'm trying to achieve is to catch the user's actions so I can display some information in a winform.
I've just read about packet sniffing, it seems interesting for what I'm trying to do.
I tryed to read packets going in and out of this app with WireShark. Every action in game was sending a few packets, but I couldn't read them as they were just a bunch of weird characters. I tryed to decrypt them using a few methods I got on WireShark's forum without success. I was asking myself, even if I could see them in Wireshark, how am I gonna do that programmatically..
There is certainly a good way to do this, as we often see bots in this game.
Considering the number of bots playing "in team", I'm pretty sure they do not use clicks, but they run something in background that sends requests.
How do you make such a bot that fight, talk, interact with players automatically?
This game is Dofus, powered by Adobe Air.
I usually program with c++ and c#, but I was wondering what's the best way to do this.
I need a kick in the right direction!
Maybe trying a tcp/ip listner control (or use tcplistner class in c#) in your c# project with the appropriate port to catch requests (& responses). Information sended could be compressed so you may want to try some standard algo.
Did you try reverse engineer the AIR app ?
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.
In Blackberry application I want to check what type of network connection is being used on particular phone, whether it is BES/MDS,BIS-B or Direct Tcp.
Is there any way to find out this?
Many applications like Jive,Opera and many more are doing this kind of check.
Please help.
The question is quite logical and I do agree with Richard as well. Though a better answer lies in the fact that there can be a logic developed which would involve Service Book parsing and making use of system listeners to check the current coverage status.
I had attempted to make one such logic once in my project which worked for me. I had shared my findings and understanding about the concept in more detail at my blog post. May be you would like to check once.
You can find my blog post here.
Your question springs from an incorrect assumption. A Blackberry could be communicating over any or all of those channels simultaneously. In fact any application may as well. At any particular time you can determine if coverage is sufficient for one of those channels, or register a listener for notification of changing status using net.rim.device.api.system.CoverageInfo.