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Voip Software Jobs

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Hourly - Entry Level ($) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
We're looking to have SMS/MMS t.38 faxing and Conference Bridge Apps made. SMS/MMS would need to send SMS replies back through email and possibly even start from an email if possible. Would be nice also to have a conference bridge as well with some basic feature sets. t.38 faxing could work with either Plivo or Voip Innovations.
Skills: VOIP Software API Development iPhone App Development PHP
Fixed-Price - Entry Level ($) - Est. Budget: $200 - Posted
Looking for a honest and professional NOC and Quality Engineer to work in Asia - Europe time zone. You should capable and experienced in Wire-shark traces, SIP traces, Create SIP Trunks of various products. Able to test the quality of the routes. Routing Adjustments Know how of Nextone (Genband Switch) Able to create voice quality reports too Responding to trouble tickets Also performing other NOC / Technical tasks regularly. You should be able to start the task immediately. Please share your previous experiences as well.
Skills: VOIP Software Asterisk Network Administration SIP
Hourly - Entry Level ($) - Est. Time: 3 to 6 months, 30+ hrs/week - Posted
Responsibilities: • Design and build new code, tools and applications related to videophone initiatives • Translate business requirements into logical, component-based technical designs • Provide ongoing code-level maintenance and support for videophone applications • Drive resolution of all issues that come up during development phase • Clearly communicate project status to key stakeholders throughout entire development process • Create comprehensive technical system documentation and provide handoff training to business and other development staff • Provide valuable input for technical meetings in regards of all aspects of VoIP development Qualifications: • Minimum 5+ years VoIP development experience, with proven ability to design mobile and desktop applications for iOS platform. • Requires 5+ years experience with SIP, RTP, TCP, TLS, UDP protocols • Requires 5+ years experience developing audio and video codecs used in VoIP, such as G.711, G.722, OPUS, H.264 and VP8 • Requires 5+ years experience developing and improving jitter buffer techniques, circular buffers, packet loss compensation, redundancy compensation, media device usage for Android, Apple iOS, OS X and Windows platforms • Experience building mobile and desktop applications for Android, Apple iOS, OS X and Windows platforms • Experience writing and debugging Android Java code using AndroidStudio IDE • Experience writing and debugging Apple iOS code using Xcode IDE • Experience writing and debugging Windows Desktop code using DevStudio IDE • Experience in using Linux shell commands to administer a remote server • Experience of understanding SIP, XMPP and RTP log files and live output • Strong understanding of the development process within a company making VoIP applications • Knowledge and proven ability to use source control systems and management toolssuch as Github, JIRA and Confluence. • Experience juggling multiple tasks, communicating to team leaders and meeting strict deadlines on development projects • Demonstrated organizational skills & ability to multi-task in a fast-paced environment with competing priorities. A generic NDA with contractor agreement is to be signed ahead of releasing access to git.
Skills: VOIP Software Apple Xcode VOIP Administration
Hourly - Entry Level ($) - Est. Time: 1 to 3 months, 10-30 hrs/week - Posted
I'm working with a radio company partner Icom, and we have purchased this SDK/API for integrating into there communication architecture. Using the SDK we can control messaging, radio id, and audio in/out. What i'm interesting in is developing a custom audio gateway plugin that would allow audio to flow in and out to SIP enabled topologies (like asterisk/Etherstack and many others) . We have the API for this. Normally I ask them to work on a transcoder, but the audio codec ICOM uses is proprietary and they only licenced us to use a hardware decoder/encoder (API's provided). The AUDIO IN/OUT and Push-To-Talk and Incoming Voice Call is exposed in the API layer, which may allow use to code to an open SIP stack. I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs). Note the current development will be in windows, but possibly moved into windows embedded later.
Skills: VOIP Software Asterisk Microsoft Visual Studio OpenSIPS
Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
Customize vicidial according to our requirement for inbound callcenter. Customization includes 1. Look and feel of the agent and admin/supervisor interface. 2. Simplified method to login to the system 3. Simplified screen to create/manage agents and extensions.
  • Number of freelancers needed: 3
Skills: VOIP Software Asterisk FreePBX JavaScript
Fixed-Price - Intermediate ($$) - Est. Budget: $5 - Posted
We will send calls to Asterix and we want to create those calls like brand new calls. and we will send them to provider. We want to manipulate all call details. All sip pockets like a new one. When we do this we want to be able to do the things below. A list of CLIs, or need to generate random CLIs based on our own rules, can we handle them. We need to map random or certain CLIs to incoming traffic so when your receive traffic, always the same CLI is assigned to the same destination. We need to regenerate your incoming calls and send them out like a new call generated from your network?
Skills: VOIP Software Asterisk FreePBX VOIP Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
We want to have Agent CallbackLogin Option (Not CallBack Extension). Is there any possiblity to implement that feature and how much it will cost for customization and time frame. (Basically What I need is: 1. Agent picks up the phone, dials 200 and listens to the agent message. 2. Enters the agent number,# and listens to a message asking about the extension (which doesn't happen in the above scenario). 3. After entering the extension,# it gets "permanently" logged in on that extension even after he hangs up. 4. He stays logged in until he logs off by doing the same procedure as above only without entering any extension number (agent number,#,#). I want to use elastix call center because of agent reporting and all Thanks and regards,
Skills: VOIP Software Asterisk Elastix FreePBX
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