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Asterisk Jobs

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Fixed-Price - Est. Budget: $ 150 Posted
1. Setup the security and call quality settings correctly. 2. isymphony complete setup 3. incoming call diversion to operator logging, (may have 2 operators) 4. correctly setup/verify approx. 10 extensions and 2 pstn connection 5. call recording schedule into dropbox or Justhost folder 6. make sure pabx run smoothly without having any security issues 7. change ivr menu options and hold on music options 8. setup on demand recording individual files to as user option to email or downloaded 9. help to setup headset’s as optimum use conditions & rectify current issues with existing trunks 10. Given full tanning including; Free PABX commands essentials Setup and use of isymphony Setup new extentions call recording Managing backup’s
Hourly - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
I need you to setup a VOIP PBX on cloud. You need to configure all the basic functionalities including and advice me with all other requirements. It should have the basic functionalities like - Autoattendant - Route calls to my cellphone after working hours - Unlimited users addition option - Live calls reports - Configure to make International calls - Call recordings After configuration you need to give me a basic training on it.
Hourly - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
I have an Elastix setup with a T1 card my D-channel keeps going up and down. I need someone to look at it and solve this issue. More information will be provided during interview. This needs to be solved ASAP. Please apply if you have proven experience doing this type of setups.
Hourly - Est. Time: More than 6 months, 30+ hrs/week - Posted
The TCP/IP VOIP Engineer will be providing functional and practical analysis related to the installation of the communication networks. The TCP/IP VOIP Engineer will be given the responsibility of Designing, maintaining and improving voice and telephony infrastructures and ensures that all are secure, available (99+% minimum uptime) and functioning efficiently. You will be able to ultilize your knowledge of commonly used communication concepts, practices and procedures to diagnose and resolve moderately complex communication issues. Coordinates strategies for defining, deploying and maintaining the companies voice and video communication architecture and its associated network connections and component hardware. The TCP/IP VOIP Engineer also manages all engineering projects for voice and video initiatives, planning technology roadmaps and configuring and optimizing all PBx telephone system and services.
Hourly - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
My company does mostly PRESS-1-SURVEY We will also be doing some outbound campaigns. I need some one who has max experience with these dialer systems. I need you to help my team in configuring it and give ongoing support in my Vicidial/Goautodialonly experts are welcome to send their proposals. If you apply for this job, please put your favorite M&M color in your reply.
Hourly - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
Need a network engineer to help plan and design a network for a medium size network. Services to include are data, voice, IP surveillance and datacenter. Please only apply if you're an expert in networking and have years of experience.
Fixed-Price - Est. Budget: $ 100 Posted
Tehnology we use: -Asterisk -FreePBX -WebRTC (Tried with Sip5ML, JsSip, Sip.js) - 4 different trunks - One stationary number (our ISP) - 4 GSM numbers Problem we have: When calling trough stationary trunk on stationary trunk we get sound delay when answering late (Delay is not always the same it depends on the time when user pick up the call) Explanation: Ok so we start off with what is working. If we make a phone call through GSM trunks on any number (GSM or Stationary) everything works fine. The important thing here is that when connection is established and call starts to progress we get response 183 (With early media) and so on... The 183 response is sent by our GSM VoIP terminal. The problems appear when calling through our ISP trunk, since provider can't fake 183 response if it is not sent to them from dialed number. When we call through ISP trunk on GSM numbers things work fine, when connection is established GSM providers send 183 response with early media...
Fixed-Price - Est. Budget: $ 200 Posted
Hi I need help with eSpeak TTS, i am having issue to play it via Asterisk, if you have experience with this , then please reply to this job posting Thanks Aron
Skills: Asterisk