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Opensips Jobs

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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: OpenSIPS Asterisk Microsoft Visual Studio SIP
Fixed-Price - Intermediate ($$) - Est. Budget: $40 - Posted
I need full configuration for solution VOIP kamailio and asterisk. Kamailio sip proxy and nat transversal, dispatcher asterisk. Kamailio need lcr database, cdr, accept registar or P2P
Skills: OpenSIPS Asterisk SIP
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
Need current Asterisk demo environment extended to use Kamailio for sip registration of two phones and one sip trunk. Freelancer will be responsible for taking a working Asterisk configuration (sip.conf, res_digium_phone.conf, extensions.conf, and voicemail.conf + any other freelancer need deems necessary) and helping provide us the ability for sip registration to be moved to Kamailio while utilizing Asterisk for backend media. Would also like to continue to register the phones (Digium D45 and D70s) utilizing res_digium_phone.conf file in asterisk if possible. If not possible will need an alternative method provided by freelancer. Key skills would be Asterisk, Kamailio, and Linux admin (specifically Ubuntu 14.04 server). Successful engagement will be with freelancer providing the following - Install script or process for Kamailio. - Steps on how and where Kamailio configuration is needed to go to test. Configuration files provided by freelancer for completed process. - Asterisk configuration files for testing new solution with Kamailio. We will take these files and do basic testing, including verifying sip trunk (Flowroute) is working, phones are registering, and backend media is still in place. Once this has been tested out fully on our side project will be deemed successful.
Skills: OpenSIPS Asterisk Ubuntu
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