Asterisk PBX Jobs
Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.
Hire Asterisk PBX DevelopersI have freepbx already installed and goip4 gateway already installed and configured. I want to configur freepbx to connect with goip4 gateway 3 lines (simcards) 3 SIP user : user 600 recive call and working with line 1 from number start with 06 user 500 recive call working with line 2 from number start with 05 user 700 recive call working with line 3 from number strat with 07 variables if user 600 dont respond he redirected to voicecall to leave a message 1 after that send sms offer 1 if user 500 dont respond he redirected to voicecall to leave a message 2 after that send sms offer 2 if user 700 dont respond he redirected to voicecall to leave a message 3 after that send sms offer 3 ps dont change any network config on goip4 gateway.
i have asterisk sip server and its working fine with any RTP i set on my system interface. but in same VM if i copy and make a new server when i try to set any new RTP its not working i dont know whats the issue audio is gone. so i need you to any how create a system where i can set any rtp as i want. like or or anything as i want. (we dont use our public IP as RTP we use any IP on our RTP like a eth0:1 interface ip is:1.1.1.1 so i will use this as a RTP) you can use any sip server or anything as you want. i just want to use RTP thats it. you can to setup this your local system or if you want i can give you server dont ask me any payment before test. if you can show me its working and audio is fine you will get payment with bonus.
need to install VICIDIAL and give support for some time .
I have already did the full process from verifying Facebook and verifying Twilio with WhatsApp with my own number and I have verified the template But I have an issue and I need someone to help me with it Error 11200
Send sms when called a Twilio number, send a link to continue in Whatsapp or sms, answer sms incoming messages as conversation on web interface, have login for customer, agents per customer and supervisor. Only agents can see active sms conversations, closed conversations are for reference and can be seen if new conversation is active with contact number, provide rports for daily, weekly, monthly conversations and time.
Hi guys, I need create and setup a IVR campaing in our Vicidial Server, to sent IVR to ours customers, and finally download a complete report with the calls status (answers, failed, no response, etc)
Hello, We are looking Kamailio and Opensips Expert to integrate the below Kamailio and opensips module We need a return Invite as per the below URL configuration Can you please help us Thank You
We have GSM GW Like Dinstar, Ejoin , SK. GW have Sims. Each Sims have own number So we have to Dial number local so need to develop some solution where GW connect some dialler and with help of dialler we dial specific number from GW sims.
There is a campaign for incoming calls, the calls that enter the agents are recorded, but the calls that enter directly to an extension registered in a softphone are not recorded, these calls are required to be recorded and can be seen from the dashboard
I need configure integration chat with clients website to Goautodial, and integrate button ClicktoDial with clients website. The idea is that on the client's page there is a chat and a call me button, so that it is integrated with a goautodial campaign and can be managed by the advisors.
The istallation of the softswitch will be used to set up a VoIP platform for a voip retail business start. The softswitch will be installed on a cloud server, and will be configured ready to start using and intergration with an app. The last step will be testing it with an app.
We require the setup and integration of Voip for odoo 15 Community version with elastix voip system.
need to make dialplan for realtime queue using mysql for asterisk system
