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Sip Jobs

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Hourly - Intermediate ($$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
Need assistance implementing an installed version of 3CX, including helping setup a FQDN (for access from outside the network, like with pbx.domainname.com) with SSL. We do have a static IP Address, and are pretty technical (but not much on the network side). You will also help us connect our phone system to outside system phone providers.
Skills: SIP VOIP Administration VOIP Software
Hourly - Intermediate ($$) - Est. Time: More than 6 months, 30+ hrs/week - Posted
We are looking for creative and motivated software developers to improve our XMPP-based front-end and back-end software stack in a long term partnership. Knowledge of various debugger tools like Firebug and IE Developers tools, Mobile UI development, Cross browser designing and developing experience. We work with cross-cultural and global teams, which require virtual communication skills, pro active behavior and the willing to think and act solution oriented. The developers must have experience working with responsive UI, cross browser front-end development. Linux and XMPP know how is appreciated. * good English skills (Both reading and writing) * experienced with GIT * fast learner, self-directed, work well with others * experienced in Ajax / JavaScript, willing to work with a new framework * experienced in Java / J2EE / JSP / servlets (Zimbra) * experienced JavaScript, HTML, DHTML, XHTML, CSS skills * Backend and Frontend experience is appreciated * XMPP, WebRTC, VoIP experience * Linux knowledge * nice to have: Java, Bash, Perl, Python * Networking (IPv4, IPv6, HTTP, TLS, network security, encryption, TCP/UDP, server-client architectures, mesh architectures, peer2peer, VPNs) Also Zimbra knowledge is beneficial.
  • Number of freelancers needed: 2
Skills: SIP AJAX AngularJS CSS
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
This is a MS Voice test bed for SIP trunk compatibility testing to allow Skype For Business to make and receive calls via external VoIP provider. ... It must allow - local Skype for Business client registration - SIP Invites will be sent to provider's external SIP proxy to initiate a call in E.164 format i.e. ... It must allow - local Skype for Business client registration - SIP Invites will be sent to provider's external SIP proxy to initiate a call in E.164 format i.e. SIP/4420812345678@proxy_hostname - Lync requires TLS+SRTP, sip provider only supports TLS - mode change required - receive inbound calls to active SIP proxy registration - our understanding is that we only need two servers: one with DC, another one with front end and mediation server roles colocated, feel free to correct this.
Skills: SIP Microsoft Lync Server Microsoft Server Skype
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: SIP API Development iPhone App Development PHP
Hourly - Entry Level ($) - Est. Time: 1 to 3 months, 10-30 hrs/week - Posted
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) . ... 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). ... 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: SIP Asterisk Microsoft Visual Studio OpenSIPS
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We need to foward on a one to one basis, any incoming call from PORT 1 on de GXW4108 as a SIP TRUNK to ring at the customer's UCM-6102; when the customers request dialout on their UCM-6102, the outbound call should be routed outbound via the SIP TRUNK to PORT 1 on the GXW4108 which is connected to the FreePBX Asterisk system ... We need to foward on a one to one basis, any incoming call from PORT 1 on de GXW4108 as a SIP TRUNK to ring at the customer's UCM-6102; when the customers request dialout on their UCM-6102, the outbound call should be routed outbound via the SIP TRUNK to PORT 1 on the GXW4108 which is connected to the FreePBX Asterisk system
Skills: SIP Asterisk CentOS FreePBX
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: SIP Asterisk FreePBX JavaScript
Hourly - Entry Level ($) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
We are looking for a Linux Server Administrator that specialises in both VOIP and VPN services. The person must have knowledge and experience with Asterisk VOIP and OpenVPN services. The job will be on a per needed bases. The administrator must have excellent English communication skills and be able to respond to technical support tickets from time to time.
Skills: SIP Asterisk CentOS Debian OS
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