Nelly codec sip rtpKazi
...roller shutter) I will build 2 doors, one is a single door to a narrow corridor in a section of the garage that will go forward shortly and then turn left into the SIP building which is adjoined to the garage. The remaining gap in the garage opening will be large workshop doors to give access to the garage. The large workshop doors will have aviation themed artwork on them. The corridor door will have company name/aviation themed images too. The inside of the corridor will be mocked up to look like a jetway bridge like: At the end of the corridor left turn will bring you into the SIP building which will contain a small aircraft cabin section similar to leading to a cockpit like https://images.app.goo.
Tafadhali Jisajili ama Ingia ili kuona maelezo.
Dear Freelancer, i would like to configure my FreePBX with Sip lines IP based Authentications,
Fazer configuração de um tronco SIP TLS no asterisk
Linhome is Belledonne's latest open source SIP SDK esspecially developed for use with SIP video doorintercom systems deployed in multiple dwelling units. Info about obtaining opensource SDK/code and required coding skills your can read here We want some minor changes in the current Linhome APP that Belledonne have published themselves in Appstore and Playstore - search for Linhome: 1) Replace Linhome logo/icon with our preferred logo/icon 2) change color scheme from orange now towards blue 3) include language support xml for automated usage of English/French/German/Dutch/Marrocan/Turkish based on the phones prime language The APP will be compiled and delivered to me running in both Ios as well as Android. We will publish the supplied
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Sip proxy sends the call to kamailio, Kamailio route the call to dispatcher1. As soon as kamailio receive second sip message (after "100 trying) from dispacher1 member sip proxy, If sip message equals to 180 or 183 from dispatcher1, Kamailio will reroute the same call to dispatcher2 Only kamailio configuration support needed. All other dispatcher or sip proxy configurations have already done by our side.
Our company has been operating for 30 years, we have had 2 logo changes in that time and are looking for a fresh new look. Planet Telecommunications core business is Telephone systems, install, service and we are also a Business grade SIP and internet service provider. Our focus for future is based on cloud based communications and carriage. You will find more detail of our business from our website including the current logo. We are looking for a fresh new leading edge logo to take us forward for the next decade, we would like the logo to be be PlanetComms instead of Planet Telecommunciations. Planet Comms can be spaced and capitilised anyway you wish. Our current colours are on our website. Looking forward to seeing some ideas.
we are an intercom company with hardware that makes video calls to an ios app. The app simply receives the video calls, and it runs on linphone client. For our iOS app, we have some users having an issue loading the video call correctly. it hangs at the loading screen and video does not load. (only some users) We would ...that makes video calls to an ios app. The app simply receives the video calls, and it runs on linphone client. For our iOS app, we have some users having an issue loading the video call correctly. it hangs at the loading screen and video does not load. (only some users) We would like this issue to be fixed. Requires a person with good knowledge and experience with iOS development and linphone SIP client. This opportunity may lead to a permanent position with ou...
Hello, We're a small company of IP telephony professionals located in Ukraine looking for a good partnership with an EU citizen or company of the same scale. We would like to establish a partnership with you for IP telephony outsourcing, SIP trunking, and DID (virtual) number provisioning. Message me if the above sounds like you. Thank you for your attention, and have a wonderful start of 2021! Dmitry
... args = parse_args() File "/usr/lib/python2.6/site-packages/redhat_upgrade_tool/", line 123, in parse_args args, _leftover = p.parse_args() File "/usr/lib64/python2.6/", line 1396, in parse_args (str(err)) File "/usr/lib64/python2.6/", line 140, in __str__ return _("no such option: %s") % self.opt_str UnicodeDecodeError: 'ascii' codec can't decode byte 0xe2 in position 1: ordinal not in range(128) You have new mail in /var/spool/mail/root [root@sr1001obv001 ~]# cd Downloads/ ...
Need a routing expert to set routes for 3 interfaces. First interface is main network using as gateway () Second interface is VoIP for phone using as gateway () Thirth infterface is direct connection to SIP using as gateway (ip ) and needs to contact
Webrtc similar chat creation with all functionalities cpanelwhm vps server instalation it's up to you to do all the installation on the server, add to my wordpress domain you designate ((((I ha...comprehensive video chat for your website, done in html5 and easy to integrate. You choose between Flash and WBERTC technology for streaming. HTML5 chat now includes real-time Webrtc multi chat CHEK URL // HTML5 chat now switches to TURN for better compatibility Since Safari 12 now accepts the VP8 codec, we have decided to use the VP8 codec for webrtc video chat. What are the advantages of VP8? better compression best quality faster We ditched the H264 codec and decided to go with Vp8. This Vp8 codec works with all modern browsers: Chromium Firefox Safari (&...
Webrtc similar chat creation with all functionalities cpanelwhm vps server instalation it's up to you to do all the installation on the server, add to my wordpress domain you designate ((((I have nothing ))) please check url ...real-time Webrtc multi chat CHEK URL // CHEK URL // CHEK URL // HTML5 chat now switches to TURN for better compatibility Since Safari 12 now accepts the VP8 codec, we have decided to use the VP8 codec for webrtc video chat. What are the advantages of VP8? better compression best quality faster We ditched the H264 codec and decided to go with Vp8. This Vp8 codec works with all modern browsers: Chromium Firefox Safari (> 12) Opera
I have a Freepbx system that has been up and running since March of 2020 with no issues. All of sudden our customers can't dial into to the SIP trunk. The ISP is claiming there is a "500" error on the network coming from our Freepbx box. Need help diagnosing this issue.
I need a sip based webrtc webphone with all basic features and audio video calling plus instant messaging is desirable but not must
Looking to develop a Par Sheet for a 3 Reel Slots Game. The game will involve 3 different Wild Symbols that not only substitute for other symbols but also multiply their values. Game needs to have a 90% + RTP and 15-20% Hit Rate Sample Paytable below Wilds can be additive
Lindoor is linphones/Belledonne's latest open source SIP SDK esspecially developed for use with SIP video doorintercom systems deployed in multiple dwelling units. We want Lindoor to use a html link to load a specific landlords logo, change colorscheme from orange now towards blue and all English text changed toward by us supplied Dutch text. The automated Provisioning assistant wizard when popping up after first start of the APP needs to popup towards the QR scanner functionality. All provisioning settings included html link for logo will be imported from the provisioning link that was embedded in the QR picture (no coding required). The APP will be compiled and delivered to me running in both Ios as well as Android. We will publish the supplied APPs in Appstore and...
...already build a unique WebRTC C++ client (Windows) that captures the host desktop & windows and streams in real-time to standard browser peers and even act as remote controller for those remote browser based participants. 2. seeking for ***highly capable C++ WebRTC expert*** to extend EXISTING functionality with advanced cool features such as: a. supporting latest WebRTC 89 library (with latest codec supports, improved video adaption features etc') b. ability to stream host mic/system audio/camera and capture and present remote A/V streams (alternative to our existing js based implementation) c. record single host entire screen + entire A/V streams locally to mp4 file or server mode (record entire room with all participants A/V/SS streams) while managing previous recor...
We are looking for an experienced artist to design and create artwork for our new Paint and Sip Studio. All paintings will need to be done using Acrylic Paint. All the specified details will be provided. We are after 15-25 different paintings and designs. We have a lot of inspiration and ideas to make this a simple process for you. Videos of you painting will need to also be created as a teaching method for our other artist. (these videos will NOT be used publicly) Thank you,
I am in need of expert to setup SIP for our TWILIO IVR system. I currently have Twilio IVR system that was designed by my previous IVR freelancer but he is unable to setup SIP feature for incoming and outgoing VOIP calls. I need someone who can set this up.
Looking for some assistance setting up a SIP trunk on our Mitel 3300 MXe controller.
Need help how to create a encrypted phone that connects to my sip server
...screen and therefore we need a speech 2 text functionality in jitsi. In the jitsi/jigasi documentation () is mentioned that it is possible to implement: "It is also possible to use Jigasi as a provider of nearly real-time transcription while a conference is ongoing as well as serving a complete transcription after the conference is over. This can be done by using the SIP dial button and using the URI jitsi_meet_transcribe. Currently Jigasi can send speech-to-text results to the chat of a Jitsi Meet room as either plain text or JSON. If it's send in JSON, Jitsi Meet will provide subtitles in the left corner of the video, while plain text will just be posted in the chat. Jigasi will also provide a link to where the final, complete transcript will be served when it
...already build a unique WebRTC C++ client (Windows) that captures the host desktop & windows and streams in real-time to standard browser peers and even act as remote controller for those remote browser based participants. 2. seeking for ***highly capable C++ WebRTC expert*** to extend EXISTING functionality with advanced cool features such as: a. supporting latest WebRTC 89 library (with latest codec supports, improved video adaption features etc') b. ability to stream host mic/system audio/camera and capture and present remote A/V streams (alternative to our existing js based implementation) c. record single host entire screen + entire A/V streams locally to mp4 file or server mode (record entire room with all participants A/V/SS streams) while managing previous recor...
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I'm looking for a VOIP/FIREWALL FAILOVER expert. We have a call center, approx 200 reps and we dial our customers based in the US via VOIP. We already have 4 different bandwidths deployed and already have a firewall instal...our customers based in the US via VOIP. We already have 4 different bandwidths deployed and already have a firewall installed which is Pfsense. Currently, we dial out from 1 bandwidth and all calls get connected from that single bandwidth, the problem is when that bandwidth goes down or we face a high latency issue, our calls go drop and we have to manually switch our calling traffic which is SIP traffic from port 5060 to another bandwidth. So, we need a solution preferably on Pfsense or any other firewall where our calls do not go drop even if the ban...
This project comes in a need to give options at the codec level, from H254 to H265, HEVC, others, in live stream. To support the encode / decode it will be based on FFMPEG and Nvidia GPUs, or similar. The Panel must work on linux environment. License system must be incorporated for software protection. More details will be provided after contact...
Hi, we are looking an developer who can develop an sip/voip softphone for android with custom screens thanks
I host an entertainment show called Trap Poetry Sip n Write where people can enjoy writing poetry while listening to classic poetry through Hip-Hop. I am looking to have the logo by the end of 2020. I have attached the flyer that I use and am looking to take the mouth part and turn that into a logo.
I need someone who can install Vicibox Server on Cloud and setup SIP account on that for inbound and outbound call center. Skills Required Linux Asterisk PBX VoIP Call Center CentOs
I need someone who can install Vicibox Server on Cloud and setup SIP account on that for inbound and outbound call center.
I am building an app where people can upload self made videos to our website. The video file is different in ios and android (mov vs mp4) I want to change every ios video (mov) to mp4. Not just by changing the suffix from mov to mp4 and. i also need to change the codec. I would like to have it build in js react native (front-end) working on expo in managed workflow. The picture is how it looks like right now, but this isnt working properly. The goal is to make every uploaded video playable on google chrome browser. I look forward to hear from u guys!
Paint Me Silly Too is a small business being launched in 2021 which sales painting supplies for a group in a box. For local customers there is also a bakery component to sale customized cakes. Slogan: Paint Me Silly Too- Party in a Box I would like the logo to be colorful and symbolize both the art and the cake component possibly popping out of a box or illustrates a su...which sales painting supplies for a group in a box. For local customers there is also a bakery component to sale customized cakes. Slogan: Paint Me Silly Too- Party in a Box I would like the logo to be colorful and symbolize both the art and the cake component possibly popping out of a box or illustrates a surprise factor. I would like these packages used for events such as: family fun; paint and sip; children pa...
We have a Cisco UC 560. We bought a Fanvill i20s Voip Door Phone. We need someone to get it connected with the Cisco UC 560. TeamViewer Available
...already build a unique WebRTC C++ client (Windows) that captures the host desktop & windows and streams in real-time to standard browser peers and even act as remote controller for those remote browser based participants. 2. seeking for ***highly capable C++ WebRTC expert*** to extend EXISTING functionality with advanced cool features such as: a. supporting latest WebRTC 89 library (with latest codec supports, improved video adaption features etc') b. ability to stream host mic/system audio/camera and capture and present remote A/V streams (alternative to our existing js based implementation) c. record single host entire screen + entire A/V streams locally to mp4 file or server mode (record entire room with all participants A/V/SS streams) while managing previous recor...
...build a unique WebRTC C++ client (Windows) that captures the host desktop screen or windows and streams in real-time to standard browser peers and even act as remote controller for those remote browser based participants. 3. seeking for ***highly capable C++ WebRTC expert*** to extend EXISTING functionality with advanced cool features such as: a. supporting latest WebRTC 89 library (with latest codec supports, improved video adaption features etc') b. ability to stream host mic/system audio/camera and capture and present remote A/V streams (alternative to our existing js based implementation) c. record single host entire screen + entire A/V streams locally to mp4 file or server mode (record entire room with all participants A/V/SS streams) while managing previous recordings...
...already build a unique WebRTC C++ client (Windows) that captures the host desktop & windows and streams in real-time to standard browser peers and even act as remote controller for those remote browser based participants. 3. seeking for ***highly capable C++ WebRTC expert*** to extend EXISTING functionality with advanced cool features such as: a. supporting latest WebRTC 89 library (with latest codec supports, improved video adaption features etc') b. ability to stream host mic/system audio/camera and capture and present remote A/V streams (alternative to our existing js based implementation) c. record single host entire screen + entire A/V streams locally to mp4 file or server mode (record entire room with all participants A/V/SS streams) while managing previous recor...
Hi Malay P., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.
Hi Ram Krishna D., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.
Olá Malay P., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.
Hi Ram Krishna D., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.
Hello Freelancers, We want to develop a Windows 10 Based SIP and XMPP client software. Basically we need a similar software like Bria. The basic functionality of the software will VoIP soft phone and XMPP chat functionality. You can use any opensource framework to start with. It should have different type VoIP codec support. We have FreePBX as VoIP server and Openfire as XMPP server. This software will work as client to them. Regards, ALSOFT
I am looking for someone to build me or give me an npm package for react-native that can be used on expo managed workflow and that can convert mov videos (filmed on ios) to mp4. Not just changing the suffix but also the proper codec so the videos can be played on google chrome.
...calling use a) Agent browser conencts to freeswitch over WebRTC, but call is sent out on SIP to our SaaS platform for call termination b) Agent browser connects over WebRTC and call is placed directly over SIP from freeswitch to SIP trunk provider. During the active telephone call we will require an API or trigger from Agent side to force the telephone call to become a conference call. Audio between the agent and the trunk will be muted for DTMF tones. A third SIP call leg will be added to the existing call connecting the audio path between the customer and third call leg parties. DTMF and audio will be active between customer and third call leg. Third call leg will be sent a XFER header in the SIP information. XFER header will be pushed with the API...
Hello dear ..we are looking solution for mobile phone make a like voip gsm gateway for android app,, call will come from sip server to android apps .. then apps will sent call to gsm dialing ,,
Freelancer will setup Asterix - SIP server for Simbank and GOIP. Please let me know about your experience first.
I need a Simple SIP Registration test app for Windows. It will mock an Asterisk server receiving REGISTRATION requests from SIP clients (Softphones or Hard Phones such as Cisco, Polycom, etc.) and it should reply as Asterisk does and it should show the entire dialog and optionally log it to a file. A simple GUI is enough, where I can create an extension (number and password) and then configure a phone to point to it and try to register (the App would get the local IP of the machine it is running on - if more than one is detected, I should be able to chose/select which ones to bind to and listen on). The listening port should also be configurable (default to 5060 but changable). The app should remember the last configuration, screen size and position, etc. I need a standalone...
Buscamos un perfil experto en infraestructura de Microsoft Teams, capaz de montar un SIP Trunk para unos 10.000 usuarios.
