Asterisk Jobs

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Fixed-Price - Entry Level ($) - Est. Budget: $1,000 - Posted
Looking for developer with the ability to integrate the agent interface, of GoAutoDial, to reflect appointments scheduled through the dialer into both a desktop application calendar and a web-based application calendar. The desktop and the Web-based applications are both built in-house. The interface for the AGENTS and ADMIN will also need altering for aesthetic purposes and to reflect our company info/logo.
Skills: Asterisk MySQL Programming openSUSE PHP
Hourly - Entry Level ($) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
We need some engineers with expert knowledge of Freeswitch/ or Asterisk. We work around the clock. Please indicate when you are available to work.
Skills: Asterisk Freeswitch
Fixed-Price - Intermediate ($$) - Est. Budget: $50 - Posted
I am looking for an absolute rockstar asterisk person that can set up our phone system in Vicidial and Go Autodial. ... The dialer must have be functionality to receive phone calls and do outbound calls at the same time for US and Canada. Person must have the following experience: Asterisk programming Vici Dial Setup Go Autodial / Go Admin Dialer Functionalities: Inbound call Outbound call Pause codes - break, training, lunch, aux Disposition - sale, hangup, disconnected, cancel, refund, deadline, DNC, Ability to adjust callback times for different timezones All calls should be recorded to server Please be committed, organized, accurate, professional, and timely with your efforts to produce amazing results.
Skills: Asterisk vicidial VOIP Administration
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Stopped working suddenly, all end clients get asked to enter a pin number Example from CLI > 0x7f64281d30f0 -- Probation passed - setting RTP source address to 24.24.191.246:35637 -- Executing [18583121997@onecentvoip:2] Set("SIP/24.24.191.246-000001e5", "CALLERID(number)=8185561985") in new stack -- Executing [18583121997@onecentvoip:3] DeadAGI("SIP/24.24.191.246-000001e5", "a2billing.php,1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- <SIP/24.24.191.246-000001e5> Playing 'prepaid-enter-pin-number.gsm' (language 'en') == Using SIP RTP CoS mark 5 -- Executing [16615335130@onecentv
Skills: Asterisk A2Billing
Hourly - Intermediate ($$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
I need to build a service using Asterisk. The service is to let users dictate recordings into a phone, but also to pause, resume, playback, rewind/fast forward playback to a new spot, and resume recording from the desired spot.
Skills: Asterisk
Hourly - Entry Level ($) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
We are looking for develop an IVR Monitoring service to help ensure our customers IVRs Services have the uptime, availability, and responsiveness required to deliver high quality of service. We need an automated IVR monitoring solution that can call, ping, and perform test requests to our customers applications on a variable scheduled throughout the day, and alert by an email an specify contact any time a problem is detected. It must a web solution able to run on a Linux server (Ubuntu o Centos), should be on Python or Node js, Angular. Our IVR Monitoring Service must includes: - Creation of a custom caller simulation script that dials and walks through your voice applications menus, prompts, and forms. - An automated network PING test to verify network connectivity and responsiveness from our server to our client facilities or datacenter. - Automated HTTP request and response content verification to test the availability and response time of our customers voice applications Web servers - GUI to admon the customers and their IVRs Services (it could be web) - Reports of each part of their IVR - Dial and network test plans, could be random or fix The tests quickly find and alert both parts of any error behavior, or load problems.
Skills: Asterisk Interactive Voice Response Python
Fixed-Price - Expert ($$$) - Est. Budget: $150 - Posted
I'm looking for someone who can create an asterisk(or other pbx) server so that I can do the following: Auto dialer 1. ... Auto dial the numbers based on their group, wait for answer: If someone answers play the mp3 file associated with that group Detect if a voicemail picks up and hangup and put the number at the bottom of list to dial later. System should be preferably Asterisk based. I will need the system to be hosted on my hyper-v server I also need advice on a sip host or equal that will allow these quantities of calls (up to 40,000)
Skills: Asterisk FreePBX vicidial
Hourly - Entry Level ($) - Est. Time: More than 6 months, 30+ hrs/week - Posted
We are a hosted VoIP company based in Canada. Our services include a proprietary asterisk-based hosted VoIP service for our customers, as well as internet access. ... We are looking for a Tier1 support representative with an excellent command of the English language (written and oral) who can provide first level support and remote training for our customers. You must be familiar with Asterisk, Polycom & Yealink products, and must be available by phone and email during regular business hours in the Eastern Time Zone.
Skills: Asterisk English Mikrotik RouterOS Network Administration