Asterisk Jobs

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Hourly - Intermediate ($$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
We are looking for an experience IT system developer to help us create an IP PBX architecture for our own needs and to be proposed to our end clients. Could you provide some examples of your work ?
Skills: Asterisk
Fixed Price Budget - Expert ($$$) - $100 to $500 - Posted
Purpose/Project Setup and configure Asterisk for the embedded platform. Asterisk setup needs to be configured for the embedded environment for the low flash memory footprint. ... Only required modules shall be installed onto the target. Utilizing Asterisk, the project needs to achieve functionalities as described below. ... Utilizing Asterisk, the project needs to achieve functionalities as described below. Target flash memory allocation for Asterisk: <7mb (excluding kernel and other packages) Target platform: Beaglebone black with Kernel 4.1.6 Functionality • Ability to support two call groups o Call group A (SIP client connection)  Can make calls using SIP trunk  Can receive calls from SIP trunk o Call group B (SIP client connection)  Can make calls using SIP trunk to only defined number (number shall be defined in the config file)  Cannot receive calls from SIP trunk • SIP notify.
Skills: Asterisk Embedded Linux VOIP Software
Fixed-Price - Expert ($$$) - Est. Budget: $70 - Posted
We have a freepbx system hosted in the cloud, latest version which comes with a VPN server. The VPN server uses OpenVPN. We would like to connect our customers to this pbx using a router with OpenWRT/DD-WRT firmware installed, their phones would then be plugged into this router and should be found by End Point Manager. We would like someone to write us a guide for us to setup the VPN on the freepbx, then setup the router, so that the phones we plug into the router appear in End Point Manager. We have already setup VPN on the freepbx and connected a desktop computer to this vpn, which runs a softphone, although we are struggling with connecting up the router and having phones appear in End Point Manager. We know this is possible but there is very limited documentation on the internet. Please only apply for this job if you have an excellent knowledge of freepbx, openVPN and the firmware (openwrt / DD-wrt). We would expect to be able to give this guide to our engineers and for them to be able to setup the VPN from start to finish.
Skills: Asterisk FreePBX OpenVPN SIP
Fixed-Price - Intermediate ($$) - Est. Budget: $150 - Posted
Hello! NEED TPO SETUP SERVER AND VICIDIAL FOR MY CLIENT asap !!!! I NEED TO RUN OUTBOUND CAMPAIGN PRESS 1 CAN YOU PROVIDE SRVER AND RUN VICI DIAL PRESS 1 CAMPAIGN FROM YOUR SIDE ? I WILL PAY SETU FEE AND MINUTES YOU SETU SERVER SETUP VICI RUN CAMPAIGN UPLOAD LEADES you need to have over 4.5 and more than 100 hours exp FOR MY CLIENT IF ACCEPTED COME TO SKYPE FOR INTERVIEW ORO14K
  • Number of freelancers needed: 2
Skills: Asterisk Linux System Administration vicidial
Fixed-Price - Expert ($$$) - Est. Budget: $500 - Posted
We are looking for a few custom reports to be made as well as a status page for our phone systems. Below is what we are looking for. We use Bicom multi-tenant system for our pbx systems and they have a api available to pull information. Call Status: If a ext is in use or not if it is ringing and the duration. Queue Stats: Hold long the avg wait or hold time is before the call is answered. Be able to list each caller that is in the queue out and how long they have been in there. Whisper: Can we record these calls? If yes can we hear all parties? We would like to pull a report on how many times whisper has been used on a daily, weekly, monthly basis. We are looking for a long term contractor to make a lot of reports also ones that can be emailed to customers to let them know what is going on without having to login to their system.
Skills: Asterisk API Development CURL json
Fixed-Price - Expert ($$$) - Est. Budget: $3,000 - Posted
Android Mobile application for calling (This will deliver the basic calling features needed for a voip service.and able to work with all SIP Standard softswitches, and support high quality Codecs Such as G.729,AMR, IAX2 Or ILBC which have high compression ratio and ability to offer Crystal clear Voice Quality). 2. Linux (Asterisk) base server (which will be integrate with the Android mobile app and ((3)) Vpn/tunnel server.
Skills: Asterisk Android App Development Java PHP
Fixed-Price - Intermediate ($$) - Est. Budget: $1,000 - Posted
Knowledge of Salesforce Programming (Apex Classes) -Will setup a Sandbox account for testing 2. Knowledge of Asterisk 11 (FreePBX) - Can provide Root Access to server (Currently we use 'Putty' to edit or add code) Call Recording Project: We need to created a section in our "Contacts Tab" in SalesForce that would be able to display matched call recordings stored on a Lylix server running Asterisk 11. ... Knowledge of Asterisk 11 (FreePBX) - Can provide Root Access to server (Currently we use 'Putty' to edit or add code) Call Recording Project: We need to created a section in our "Contacts Tab" in SalesForce that would be able to display matched call recordings stored on a Lylix server running Asterisk 11. A screenshot is attached to this posting. ... The phone recording files are stored in the default folder. /var/spool/asterisk/monitor/ Currently we access the recordings through the "FreePBX in a flash" control panel" Code Rules: 1.
Skills: Asterisk CSS FreePBX HTML
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
we are looking for VOIP Engineer/Administrator who can work with Kazoo, Kamailio, FreeSwitch, CGRates. complete configuration and modification to make full class 5 switch and also experience in network and app development for cloud base VOIP system o Voice and Video Telephony o Instant Messaging (IM) and Presence o Voice/IM conferencing o Interactive Voicebox and Voice-to-mail o Fax-to-mail, Webfax and Fax Clients o Serial and Parallel Call Forking o Periodic, time-based Call Forwarding (CFU, CFB, CFNA, CFT) o Inbound and outbound Call Blocking and anonymous Call Rejection o CLIP/CLIR, DDI, DID and Extension Dialing o Signaling over UDP, TCP, TLS, WS, WSS o DTLS-SRTP transcoding for WebRTC bridging
Skills: Asterisk Freeswitch SIP VOIP Administration