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Adding Asterisk WebRTC SipJS to app

Looking for someone with skills in:

ReactJS

Redux

PostGresSQL (Sequelize)

Pro in Asterisk and WebRTC

Excellent Web responsive developer

COMMUNICATION IS IMPORTANT TO ME, GOOD, RELIABLE COMMUNICATION - !IMPORTANT

WEBRTC WITH SIP via SIPJS, JANUS, or SIPML5 - !IMPORTANT

SKILLED AT MAKING DIALPLANS IN ASTERISK AMI AND ASTERISK AGI - !IMPORTANT

NEED SOMEONE WITH VERY GOOD ENGLISH - !IMPORTANT

The application is in the process of being built. I need someone to take care of the additional tasks. I need someone who can dedicate their time and provide clean and organized code.

1. "Build in call routing into the groups. Inside the SMS Group settings, when a call comes in we need to be able to decide if the call will:

- Be forwarded to ring the last agent that spoke to the customer when the conversation is still open status

- Be forwarded to ring the entire group for the first to answer

- Be forwarded to another group

- Be forwared to a particular user

- Be forwarded to an outside number

- IVR

At any point during the call, any user in the group should be able to see a join button and join the call ""room"" and join the conversation. Calls should be "rooms". There should also be the ability to make a third call to join the room (an external number).

The group should also have a timer before the call is forwarded to an answering machine which will have to be a setting in the group. Should also deduct their sms count or money depending if they finish their sms count."

2. Add in error checks that send me an email when something is wrong or the recommended error notification for tests

3. Reports:

- Message Summary Report

- User Summary Report

- Broadcast Summary Report

- Call Summary Report

- Download Logs

4. Set up outbound call where it automatically uses the right number but also have a place to choose which number to call with

5. Create lists using current contacts and to be able to upload contacts to list

6. Ability to make campaigns to broadcast from lists. There will be no list tab, but a broadcast tab only. They will be able to pick a list and choose broadcast and choose an SMS group. The numbers that will be used will be the numbers in that sms group where the returning messages will come in. There will also need to be a log/report of all broadcasts.

7. Ability to choose to purchase Batch numbers by State

技能: JavaScript, HTML, PHP, 网站设计, CSS

查看更多: issabel webrtc, freeswitch webrtc, asterisk 16 webrtc, asterisk server configuration in centos step by step, sip js freeswitch, webphone for asterisk, webrtc softphone, sipjs server

关于此雇主:
( 0个评论 ) Santa Clara, United States

项目ID: #25611989

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