OpenSIP / Kamailio with CPS control

Customer has a few dialers using asterisk and sending call to a single provider. Combined call can goes up to 100-200 calls per second, but provider only provide 30 CPS (Calls per seconds)

therefore, we need to have sip server that can

1. collect and gather all the calls from multiple dialer.

2. to control and send calls in batches, 30 cps. (user define number)

3. if call exceed 30 cps, it will be stored in buffer and send out on a First in first out basis.

4. must be able to send calls to multiple provider with different user assign CPS

5. able to do media bypass

Навыки: SIP, Asterisk PBX, VoIP

О клиенте:
( 23 отзыв(-а, -ов) ) Georgetown, Malaysia

ID проекта: #33057882

3 фрилансеров(-а) готовы выполнить эту работу в среднем за $893


hi, I am skilled voip developer. I have ready solution for you(opensips + rtpengine + web interface) , developed by me. Only one thing is complex here, it is delay for calls. There is possible to do it with various way Больше

$2400 USD за 7 дней(-я)
(78 отзывов(-а))

ratelimit module of kamailio is the solution That can help acheiving this task Cost is 20 USD/Hour Regards Wael

$140 USD за 7 дней(-я)
(10 отзывов(-а))

Hey, I am having more than 10 years of experience in Asterisk PBX and have confident enough to work with you on this project. I have gone through your requirement, and you are looking for a help in OpenSIP / Kamailio Больше

$140 USD за 7 дней(-я)
(10 отзывов(-а))