Target Phone Number Conference Auto Dialer - Asterisk / VOIP

Target Phone Number Conference Auto Dialer

Skills required: Asterisk / Trixbox / PHP / Web Meetme / Asterisk-Skype Integration. Please only submit a bid if you consider yourself an Asterisk Expert and have at least 1 year working experience with VOIP by Asterisk/FreePBX/etc.

General Description:

This is a system is designed to solve a common problem when getting ahold of your elected representative (radio talk shows, cspan, or any telephone number), BUSY SIGNALS. This is a system where registered “conference” participants may join together in a traditional conference call where they can openly communicate and a moderator has full control over the call. The difference about this conference call is everyone is there not to speak with themselves, but with someone else privately, one by one (in order). In this example, they are participating to speak to their Senator John Smith, so they arranged this conference call to have the maximum recognized impact.

A moderator defines a “auto dial number” associated with the conference call (The senators phone number). When triggered, the system will begin to use a % of all available lines to dial the phone number defined (reserving lines for new participants ). The system will likely encounter many busy signals upon dialing, the system is meant to detect a busy signal and redial over and over until a ringing line is detected (a list of auto dial numbers per conference also supported). When a ringing line is detected an user (FIFO) is transferred out of the conference call and conferenced onto the now ringing line. The phone conferenced/paired call is now being recorded as well. The user will wait until someone answers the ringing line (or they are put back on hold), speak to the other party and when they are done (or a dropped line is detected) the user will press a defined sequence of phone keys (##). The user will then be transferred back into the original conference room. Meanwhile, simotameously the system continues to use additional lines to dial the target phone number, redialing when busy. For each ringing line detected a user is taken out of the conference room, and then put back in when their session is completed. This process continues (during which additional users may join the conference who will be given first priority over repeat previous users on the line) until the conference end time or the moderator manually ends the auto dial process.

How it works:

1) Moderators schedule conference calls specifying the start and end time, max # of participants, name, description, target auto dial phone number(s), and various conference preferences.

2) Anonymous will browse the website for available “Conference Calls” on a Calendar.

3) Anonymous will register on the website as a user.

4) User will RSVP to participate in a conference call, receiving email reminders and the conference/meeting ID#.

5) On the day of the Scheduled conference, near the time of start (after receiving reminder “about to start” email) the User will dial phone # or skype into phone system. We want to uniquely identify each User who participates, therefore we will implemet one of the following IVR based authentication options per your recommendation:

a. [no meeting/conf# id required] Greeted by auto attendant, recognizing them by caller requesting PIN# to process login. If no caller id, prompt for phone number as login in as well as PIN#. Once the user is logged in they are automatically entered into the next starting conference for which they have RSVP’d already.

b. [meeting/conf id# required]Greeted by auto attendant, they are asked to e[nter the meeting ID#. Then they are asked to enter their RSVP’d phone number (unique numeric id) and PIN#.

6) The user waits in conference room until the meeting start time (or moderator manually starts the auto dial process).

7) Moderator may pause auto dialing at any time, end the conference, view, kick or ban users participanting, as well as users who haven’t yet signed into the call.

a. A feature where the moderator may trigger the system to call individual users who RSVP’d but are not on the line (provided the user specififed their phone number), automatically bringing them into the conference if they desire (1 yes, 2 no IVR)

8) The system begins the target phone#s auto dialing process using many different lines, and for each ring tone detected brings a user out of the conference pairing them onto the ringing line, recording the call.

9) Silent Reserve Mode: If this mode is enabled for the conference then, when a ringing line is detected but no users are available to pair the line it will be placed on a silent hold until a user becomes available or the line is dropped. This is also useful for lines that have a long average hold time, allowing participants to join the conference late, still having their seat reserved.

10) User “hold my place” Mode: Each user once joined into the conference may choose to hang up, reserving their place in line. When a ringing line is detected for them the system will automatically dial the phone number associated with their account (or specified/overridden). This is a feature not available to all users, and must be enabled on an account by account basis.

11) Recorded phone calls are posted onto the website in their raw unedited form, associated with the users profile and history.

Evner: Asterisk PBX, PHP

Se mere: target phone number, asterisk auto dial conference, asterisk auto conference, auto dialer asterisk php, dial telephone conference asterisk, auto join conference asterisk, pin auto dial calendar, voip dialer conference, phone automatically dial pin, join conference asterisk calls, trixbox schedule conference call dial numbers, join conference using auto dial asterisk, rsvp trixbox, skype auto dial triggered, numeric conference, form auto calls conference, skype pin dialing, asterisk reminders conference, asterisk dial phone automatically, asterisk conference auto dial

Om arbejdsgiveren:
( 19 bedømmelser ) brooklyn, United States

Projekt ID: #609823

3 freelancere byder i gennemsnit $750 på dette job


Kindly Check my PMB

$750 USD in 6 dage
(14 bedømmelser)

please check PMB.

$750 USD in 15 dage
(0 bedømmelser)

It is more likely to be a queueing system with a limited window time. ITS is s team of expertise in VoIP, Software Programming and Network design. We have successfully completed couple of conferencing system. This proj Flere

$750 USD in 10 dage
(0 bedømmelser)