Asterisk Call API
$30-250 USD
Pago na entrega
Hello,
I need an application like this:
http://ip/[login to view URL]{CALLERID}&call={CALL}
If I request this web address, Asterisk will:
1-) Call {CALL} via {CALLERID}.
2-) After CALL answers, send call status (busy, rejected, answered etc), duration and recorded soundfile path to our REST API service.
- Need support over 50 calls at same time.
- I have a server at digital ocean. Asterisk is not installed.
- I have a SIP account allows caller id spoofing.
Very simple.
I will give you server information.
Thank you.
Project plan has been changed:
This project will use twilio & asterisk.
This project will use click2call technology with custom caller id's.
http://ip/call.php?callerid={CALLERID}&call={CALL}&callerid2={CALLERID2}&call2={CALL2}&soundfile={SOUNDFILE}&id={uniquecallerid}
If success, echo "true", if not echo "false".
{uniquecallerid} is the id of operation, will given by us. If this id has been used before, echo "false".
1-) system will call {CALL} with callerid {CALLERID}, after answer, will invite {CALL2} with {CALLERID2}.
2-) If {SOUNDFILE} has been defined, play this soundfile to first call, after answer, before transfer.
3-) If {CALL2} is not defined, just wait for 1 minute, then hangup call. If soundfile has been defined, play this soundfile also.
4-) After call ended (both CALL2 defined or not), send call report to our server like this:
http://apiserverip/report.php?id={uniquecallerid}&call1status={busy|answer|noanswer}&call2status={busy|answer|noanswer}&duration={secs}&record={path/of/record.mp3}
5-) If mp3 is not possible by default, convert it before.
Also Installing required server softwares has been included in this project.
ID do Projeto: #18153788
Sobre o projeto
Concedido a:
This is a click to call using PHP, I have that system already working and willing to provide you a quick demo
10 freelancers estão ofertando em média $164 nesse trabalho
hi yes, i can do this for you. check my feedback, i am best voip expert here. please note, server should be dedicated or vps with enought cpu power. On low-end vps can be issues with sound(choppy).
I can do it, I have 9 years of linux and voip experience and I am a digium certified, I am sure I can do it thanks
Hello I have done similar works in the past. I have 10 years experience with asterisk .please contact me to discuss more in the project. Thanks
