Freeswitch Jobs

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Hourly - Intermediate ($$) - Est. Time: More than 6 months, 10-30 hrs/week - Posted
The project uses: Over 30 "WhApps" (all written in ERlang) CouchDB RabbitMQ AMQP Message Bus Kamlilio SIP Proxy Freeswitch SIP Media Server Kubernetes (scalability) Docker (containers) Here is a link to the github repo for Kazoo 3.0: We will be contributing all of our code back to this community, and I would like you to manage the pull requests and use the developer forum at!
Skills: Freeswitch AMQP Docker Erlang
Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
We need to dev our mobile app for android using webrtc integrate with our freeswitch-kamailio server.
Skills: Freeswitch VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
we are looking for VOIP Engineer/Administrator who can work with Kazoo, Kamailio, FreeSwitch, CGRates. complete configuration and modification to make full class 5 switch and also experience in network and app development for cloud base VOIP system o Voice and Video Telephony o Instant Messaging (IM) and Presence o Voice/IM conferencing o Interactive Voicebox and Voice-to-mail o Fax-to-mail, Webfax and Fax Clients o Serial and Parallel Call Forking o Periodic, time-based Call Forwarding (CFU, CFB, CFNA, CFT) o Inbound and outbound Call Blocking and anonymous Call Rejection o CLIP/CLIR, DDI, DID and Extension Dialing o Signaling over UDP, TCP, TLS, WS, WSS o DTLS-SRTP transcoding for WebRTC bridging
Skills: Freeswitch Asterisk SIP VOIP Administration
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We are developing VoIP based calling system and prepared web widget to register it as a SIP client using websockets. Few months back chrome and other browser had deprecated media and other internal APIs on non secured origin. And due to which we have rebuild our widget using wss, but we are facing some issue in it. We are using Kazoo( as a VoIP server and in this kamailio is used for the device registration and initial call setup. So we need a VoIP expert who can debug and solve issues we are facing with the secured websockets in kamailio. We have enabled TLS and also configured proper SSL certificate in the kamailo. Also able to register and make outbound calls using wss, but facing issue for the incoming calls. We are looking for some experience person who can help us solve this issue and have good knowledge of SIP and web-sockets can apply for this Job.
Skills: Freeswitch SIP WebRTC
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
- prepare a routing plan for outbound traffic (add gateway) - prepare incoming that no work on Newfies (beside what we develop) - manage positive marging and then sell only the route I am not loosing money to. - enable an account for a client with prepayment for traffic - take off all Fusion logo and name - impose and control the outgoing number of call/min of the dialer - randomize the duration of the outgoing calls
Skills: Freeswitch Django Python SIP
Hourly - Entry Level ($) - Est. Time: More than 6 months, 30+ hrs/week - Posted
. • 8+ years as a System Administrator, preferably in a client services environment. • In-depth experience with Asterisk VoIP, Freeswitch, and SIP • Expertise in PostgreSQL. • Experience with automated regression testing and quality assurance tools. • Solid understanding of efficient and secure networking, including IP protocol, firewall administration, software load balancers, routers, etc. • Ability to troubleshoot hard/software and discuss issues with various stakeholders and create clear reports on issues and solutions. • Experience programming with Ruby on Rails, Perl, PHP, and Shell Script. • Solid experience with planning, performance tuning and execution of environment scaling. • Detail oriented with excellent analytical and diagnostic skills.
Skills: Freeswitch Asterisk Linux System Administration Network Administration
Hourly - Expert ($$$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
Looking for experience telephony application developer. Please do not reply if you've only worked casually with these technologies - we really want someone that has worked on a number of projects or at least one decent sized project with numerous features that match what we're looking for below (in other words, if you believe that you can take on this job and learn as you go, please do NOT respond - we really want someone that we can rely on that can answer 95% of questions without needing to Google for an answer). I realize this may sound very direct, but I just want to be clear on our expectations - we really are nice and fun to work with, we've just had too many experiences with amateurs pretending to be experts! To begin - we have an existing telephony/IVR application and are looking at creating an entirely new application. This new application has the requirement for additional features that we do not currently have expertise in so we want to add this expertise to supplement our in-house team. Please see attached document for additional details. This is the first of a number of projects in this category that we are looking for expertise on so our intention is that this is the beginning of a longer-term relationship. We can do this project as rate per hour or fixed rate (once we have further defined the scope in relation to your skill set).
Skills: Freeswitch Asterisk Interactive Voice Response Microsoft Lync Server