The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [đăng nhập để xem URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to work. Nee...
"Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot
... Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we
I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [đăng nhập để xem URL] [đăng nhập để xem URL] The 3cx gateway for skype is running only windows 7. and needed skype old version
...of project. For ongoing project, and future project we search for a developer with skills and experience in these areas. Keywords: Kernel, Alsa, audio drivers, I2S, Drivers, Codec, DSP, etc Please note this ad is written by a project manager and is not detailed. If you think you are good in doing audio on Linux system, please send us a note and a senior
...TAS5825M (codec) with the IMX8M soc by either developing an alsa card driver or using the built-in simple-audio-card driver. 2. Develop a Linux driver for the si4688 DAB/FM radio chip. This would ideally be a Linux mfd driver that provides a control interface to userspace and also provide an Alsa capture codec. And also integrate the codec driver with
...*** - Autoconfiguration - All Basic Features of MuMuDVB - Trancoding Option (FFmpeg & FFserver) - Descrambling Options - Status monitoring - Output :MPEG TS-over-UDP/RTP supporting SPTS(multicast) - All options need to be avaliable for configuration locally Thru scroll down selections. - Support Video/Audio Transcoding/ Transrate any-to-any
...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...
I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call