SIP based service DTMF to Action URL

Job ID: 33616837

Budget: $750 – $1,500 USD

Looking to create a service using any existing SIP based solution.

Can be done on any OS, Linux or Windows. It may use open source solutions
or SIP SDK, preferable solution is on Windows OS, but not limited to it.

Requirements:

SIP Trunk, In/Out with IP or registration authorisation;
Dial Plan, simple to allow/disallow prefixes;
Black List for incoming calls to reject call based on Caller ID;
Auto Attendant account for incoming calls with the audio prompt;
DTMF detection on caller side for auto-attendant to forward calls, and callee side;
Action URL GET method to send depending on matched DTMF;
Database to store data can be any;

Web pages:
* Auto-attendant with the ability to upload WAV file (add/edit);
* DTMF to forward calls from AA
example-
301 -> 4163421690
302 -> 6477271077
etc.
* SIP Trunk configuration;
* DTMF to Action URL, and DTMF sequence to set a blacklist action
example-
4523 -> http://google.com
4576 -> http://yahoo.com
etc.
* Black List page where record can be removed/add;

Sequence or Call Flow:

1. Caller calling to a number, call goes to the trunk. answered and got an auto-attendant;
2. Caller uses his keypad to send a DTMF;
3. If DTMF matched to one of the record in DB, call is forwarded to the callee number
i.e. if used 301 then call forwarded to 4163421690;
4. Callee answers the call and can talk to the caller;
5. Callee can send a DTMF during the call, and depending on matched record action can be as
a) if DTMF matched Blacklist, then Caller is blacklisted and call disconnected
b) if DTMF matched an action URL, then the URL is executed
i.e. if 4523 then url http://www.google.com is sent using GET method.

6. Call disconnected on any party hang up;
7. If Caller is blacklisted, the service will decline the call;

Note:

All existing solutions like FreePBX, Vodia, PortSIP etc. can provide this tupe of service,
with the one exception, no-one passing DTMF from Callee's side, here is the issue.
So, please be careful when reading and be sure that you understand the task in full.
Related categories: Asterisk PBX VoIP SIP