Automated Outbound Missed Call Dialer

Our project is similiar to all those automated outbound dialers, except we need some customizations due to we are only looking to send missed calls (outbound calls to be disconnected after 1-9 second of rings) - we do not mind if you work with any existing platform or configure any other available software likes IVM, Asterisk etc to the task as long it can fullfil the following requirements:

1. We are looking forward to send massive amount of missed calls using hundreds and thousand of ports/concurrent, therefore- your solution must be server friendly, we prefers it to be linux/web oriented but won't mind if its windows based application as long it can work with 1000 sum ports with 128GB windows server (if more then better).

2. Multiple "Caller ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple callers id which to be round-robinly used for outgoing calls, if caller-id is defined for the sip-account then will only send calls using that caller-id instead.

3. Re-Dial Busy/Unavailable numbers, only calls which has ranged defined amount of times will be marked as "Completed/Done" others will be re-tried x amount of times (we will define the amount when creating the campaing) after x hours passed (like if 7 amount of retry after 24 hour passed provided then it will call and see if rings every 1 day for 7 days)

4. We would be able to define "What to do, when a called is Picked" means, if a call is connected, we would be able to define if the system is to disconnect the call immidiately or play an audio instead, if immidiately to be disconnected then calls will be dropped as soon picked before the ring period is over, otherwise will play an audio file and disconnect the call after.

5. Process "Incoming Calls" when an incoming calls arrived (if enabled in the SIP/IAX connection when configuraing it) it will be processed in four methods: (One) Receive the call & play an audio file then disconnect it (Two) Send a busy-tone [cut the call] then do visit an website URL to send a SMS to the caller-id using HTTP API calls, like [url removed, login to view][caller-id]&msg=hello! (Three) Combine both option one and two, means receive the call and play the audio and then disconnect the call and send the sms by visiting the URL (Four) Simply disconnect the call [send-back busy-tone] and store the number in a text file so we can do whatever we wish to later by extracting the data from the text file.

6. Optional - if possible, we would love to have Skype intergrations like softwares as ([url removed, login to view]) does, we would only accept this functions if multiple Skype Accounts can be used to send calls, otherwise this option is not needed.

7. Generate Reports - provide a complete reports of how many numbers has been successfully ringed, how many numbers to be re-tried, how many calls has been picked-up and how many seconds/minutes was the duration of picked-up calls, etc etc. We would also like to extract the numbers that was successfully ranged and numbers that is on re-try list and also numbers which we coudn't connect even after re-tries.

8. Send calls to random numbers - we will define the prefix like +880171 then provide how many numbers to insert after the prefix like Bangladeshi phones has 12 numbers so as we are providing +880171 additional 6 number is either to be generated in sequencing mode or random mode (as defined) then start calling, if sequencing then the first number will be +880171000000 and last no is +880171999999 while if random then numbers like +8801713450147 and +8801717888946 will be generated randomly and called but in that case, the system shoud'nt dial the same number twice, means even if generated then it will only use one number once.

This project has been undertaken before, I have copied the request as it is exactly what I am looking for

Beceriler: Asterisk PBX, Linux, PHP, Sistem Yöneticisi, VoIP

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İşveren Hakkında:
( 0 değerlendirme ) Unknown

Proje NO: #16239573

Bu iş için 17 freelancer ortalamada €614 teklif veriyor

in %bids___i_period_sub_35% gün içinde555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(75 Değerlendirme)

I have 6+ years experience and I have done more then 50+ projects in my [login to view URL] note I am full feldged developers and work for 17 hours in a day.I am ready to start the project kindly open the private message box Daha Fazla

in %bids___i_period_sub_35% gün içinde542%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(71 Değerlendirme)

hello, we have 8 year experience with Missed Call Dialer solutions. Also good experience with vicidial, freepbx, a2billing, voicebroadcast, goautodial, osdial. we already modified report & gui as per customer requir Daha Fazla

in %bids___i_period_sub_35% gün içinde750%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(35 Değerlendirme)

I have read your project details. I have to ask a few questions. Can you please message me via chat so we can discuss all the details to elicit all the requirements and hence start the development? I will showcase al Daha Fazla

in %bids___i_period_sub_35% gün içinde555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(10 Değerlendirme)

Hello Sir, I can able to configure all your requirements with Asterisk, FreePBX. And I have already done 50+ similar projects. Please go through my customer reviews and feedback in past & years.

in %bids___i_period_sub_35% gün içinde310%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(44 Değerlendirme)

Hello, We have more than 9 year of asterisk in [login to view URL] have develop real time application for web and appWe are using Centos,Linux server and codeigniter framework for web [login to view URL] check our few work sample.Y Daha Fazla

in %bids___i_period_sub_35% gün içinde1111%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(9 Değerlendirme)

Dear Sir / Mam, Hope you are doing well. Thank you for giving us opportunity to present an approach for your esteemed project requirements. Inextrix is a Software development and consulting firm in Ahmedabad (Gu Daha Fazla

in %bids___i_period_sub_35% gün içinde520%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(10 Değerlendirme)

i have developed many projects based on asterisk, ivrs, autodialer , call center solution, office communication, branch calling and more

in %bids___i_period_sub_35% gün içinde400%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(7 Değerlendirme)

I have made kind of project on asterisk and opensips basis system, if you have correct sip provider i can handle this task smoothly. In previous job have not any gui, i will install all things in beginning and developp Daha Fazla

in %bids___i_period_sub_35% gün içinde722%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(2 Değerlendirme)

Hi Robert here. I read your project description carefully and i have to say you this. Should you want to go through with a good product which can do the job, your budget does not meet the expectation. I placed a highe Daha Fazla

in %bids___i_period_sub_35% gün içinde1111%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(3 Değerlendirme)

Hello, I am Nour, Web/Mobile developer & designer with 8 years of experience, there is some APIs that we can use for the automated outbound dialers and all the requirements you mentioned are possible to achive with Daha Fazla

in %bids___i_period_sub_35% gün içinde777%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(1 Yorum)

here you can see our missed call solution : [login to view URL] Benefits of Missed Call Solution: [login to view URL] customer/caller is not charged for the missed call [login to view URL] is user-friendly as it is Daha Fazla

in %bids___i_period_sub_35% gün içinde383%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

Ya he realizado varios proyectos por el estilo. Puedo aportar experiencia y algunos datos para lograr el objetivo buscado con el discador. Luego de haber determinado el alcance y el entorno de trabajo de la fase 1. Daha Fazla

in %bids___i_period_sub_35% gün içinde566%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

Hi,   I have gone through your project details and I assure I can do it  as  i got all the skills  required for this project. I have already completed a lot of similar  projects in the past so it is my pleasure to work Daha Fazla

in %bids___i_period_sub_35% gün içinde250%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

I am certified asterisk,linux,Sip engineer and working with this types project last 7 year We know how it will work.

in %bids___i_period_sub_35% gün içinde555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)
in %bids___i_period_sub_35% gün içinde777%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

hi, we have good experience with Asterisk based voip solutions like inbound & outbound dialers , missed call dialer etc... so, please give us chance to work with you. sk ype ID : asterisk2voip ontact no : 962 402 29 Daha Fazla

in %bids___i_period_sub_35% gün içinde555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)