Asterisk FreePBX Special CODE

$30-250 USD

クローズ

投稿日:

約7年前

$30-250 USD

完了時にお支払い

Need to be able to live transfer call with originating caller id. Part 1. Agent will call Customer, and tell customer we can help with service and will transfer to Specialist. Part 2. Agent dials specialist while customer still on line. Specialist answers, we make introductions between customer and specialist, and disconnect from call. Allowing Customer and Specialist to continue conversation. Part 3. Very Important: When Agent is dialing specialist, Caller ID must show Customer Caller ID. Part 4. Very Important: Must be able to have access to entire call recording, from initial call from Agent to Customer, Agent to Specialist, then Specialist and Customer full conversation even after Agent hangs up. Thanks! Happy Bidding Rob

Asterisk PBX

プロジェクト ID: 13446715

プロジェクトについて

6個の提案

リモートプロジェクト

アクティブ 7年前

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この仕事に6人のフリーランサーが、平均$312 USDで入札しています

@meral

hi i can do it, but no way 100% garant cid for 3 will be delivered. you need sip contract with ability of cid change, but sometimes it just not work(depend of country) no issues with record

$568 USD 15日以内

4.9

(143 レビュー)

8.1

@rsatinder

Hi, I can get this done. We can use either queue module or conference module to develop this feature. I am available for any discussion if needed, please let me know. Thanks & Regards, Satinder

$155 USD 3日以内

4.9

(14 レビュー)

4.9

@bhruguios

Hello I have developed call center usign asterisk before. IVR, Media server, Agent server, Call monitor server, CDR. Where I have developed Queue/Agent/customer scenario using asterisk, mysql so I think I can complete your task. let me know if you need more info.

$200 USD 4日以内