Objective is to install and configure Asterisk and to create a script that will test telephone numbers.
I want to be able to use a super-simple web-ui to upload a csv file containing a list of numbers to test to the Asterisk server.
You would need to create the web-ui (to be hosted on that same server) as well. Again this can be SUPER simple.
No fancy graphics, no thrills.
The format of the CSV would be:
number(in E.164),mode,pin,destination
And "mode" would be either "1stage" or "2stage".
On that UI I want to be able to either tell the system to start testing right away (and only once) or to tell it to start testing in a (very limited) cronjob fashion. Like for example start once a day at 4am, or every Friday at 2am.
There are two types of tests to be performed.
1stage:
Just call the number using a SIP outbound trunk and this number would (if everything works) then send the call back to the system through a SIP inbound trunk. As soon as the script sees the incoming invite it should answer the call and send a random string of DTMF tones down the line.
The script should detect the DTMFs received and hang up. If there was no error (DTMF sent = DTMF received) then the number is to be marked as "ok".
If there was an error the type of error should be noted. So for example the call might have not gone through (no answer or unknown number) or the DTMF sequence was not received correctly.
Depending on the settings in the UI the script should run those tests in parallel (up to 30 tests in parallel).
Once the end of the list of numbers is reached the script should send a report via email (external SMTP). The report will show all erroneous numbers first (incl. error info) followed by all numbers without issues.
2stage:
Is very much like 1stage but in this case the dialed number would not automatically send the call back to the Asterisk server but instead would answer and present a voice prompt. The script would send the associated PIN (from the CSV row) via DTMF and would wait for a secondary dialtone.
Once detected the script would send the "destination" number via DTMF. That number would then send the call back to the Asterisk inbound trunk and the rest of the test is exactly like 1stage mode.
In order for the DTMF recognition to properly work the script should only allow G.711 a-law or u-law calls.
This is not rocket science but nevertheless I want to make sure that you completely have read (and understood) the requirements before placing your bid.
That's why I will ignore any bid that does not start with the sentence "Yes, I have!".
Yes, I have! I made similar auto-dialer solutions.
Please check my project reviews here -> http://www.freelancer.com/u/ivan381eu.html
Also, please check private message I sent you.
Yes, I have. I have done such projects and I'm ready to do it and you will have the solution ready and tested in a week. You are free to check my profile and see the customer feedback, almost all of my projects have been done on time. I offer a week of free support. I'm ready to provide technical details how it will be done and how it will work. Asterisk supports three way of dtmf: inband, rfc2833 and info. You mentioned about inband way, I would recommend to test all of them and use what will be better for you. Based on my other jobs I recommend to use rfc2833 instead of inband as you will get a outband dtmf and it will works everytime and the dtmf will be received same as you sent.
Yes, I have!
and i have developed many system like that.
i have developed many custom asterisk application, web application for many type of service.
let start by contact me and you will got it soon
thanks,
voipmanvn
Yes, I have!
I have ready solution ,and demo for u ,I also did custeromize job ,which support multiple-questions .
also my solution support sub account (but I can disable it to simplify the solution)