从SIP面板中找到100家从事房屋建筑的公司 如果事情不明确，请写信给我，我会尽力解释。 在中文里，我通过谷歌翻译写作。 我的母语是俄语。
Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.
VoIP Video IM等即时通讯功能。 支持Android，IPhone，Window Phone等客户端应用。 熟识SIP Webrtc XMPP。 提供Demo。
主要指导elastix asterisk 的安装配置工作，熟悉Asterisk的模块（包括体系结构，配置文件，Log日志,History）,能重新打包ISO文件
...3、要求操作简单，点击“通话”功能按钮即可实现通讯； 4、由于即时通讯双方可能位置很远，因此必须经过互联网传输，需要中转服务器进行语音媒体的中转。中转服务器可以采用成熟的第三方软件或者协议，但是在项目完成后可以交付给我们； 5、发送方网络的最低可能是ADSL 2M。但是延时不能超过1S； 6、语音质量要高，不能有杂音、回音等问题，需要可以用于商业行为的那种； 7、需要有可供调用的组件和调用方法，我公司可通过调用方法进行上层功能的开发； 8、需要提供所有源代码和演示版本（演示版本的运行平台包括android、IOS）,以及可交付的中转服务器； 9、并不需要完整的语音通讯软件，只需要能够实现语音传送、播放的功能模块即可。 10、演示内容：A用户点击“通话”按钮，中转服务器可通过设置好的IP地址，找到B用户，并发送通话请求。B用户接受请求后，双方可以进行通话。 11、你可以采用第三方组件，例如SIP，但是需要告诉我们是哪家的，效果如何。因为考虑到我国网络问题，所以建议是自己搭建服务器。 12、我们已经开发的软件可以在安卓、苹果商店找到，叫IEMAKER，通过安卓盒子，可以实现直播，但是目前还不能实现声音直播，因此，需要开发这个模块，用于该软件。
I'm seeking a skilled artist to conduct a 'sip and paint' workshop in a larger art exhibition context with more than 20 participants. Details: - The participants are novices in painting and hence, ideal candidate should be proficient in teaching beginners. - The key focus of the workshop will be to enage the guests, teach them few basic painting skills - oil, acrylic, landscape or fluid art. Skills Required: - Excellent painting skills - Strong teaching abilities, especially handling a large group - Prior experience in conducting workshops So, if you're someone who has experience in teaching novices the beautiful art of Landscape Painting, you could be just the one I need. Looking forward to work with a patient and engaging artist who could make this an u...
...persona con experiencia configurando la libreria de Javascript SIPML5 y asterisk. Tenemos todo instalado y configurado. El softphone web se registra al asterisk, emite y recibe llamadas, pero cuando se atiende la llamada no transmite el audio. El servidor tiene instalado una VPN. Cuando el usuario se conecta a la VPN, entonces funciona el audio de la comunicación pero cuando no se conecta a la VPN entonces vuelve el problema del audio. Se requiere que el softphone funcione sin VPN, solo por internet. Version asterisk 18 OS: Ubuntu Server 14 +++++++++++ A person with experience configuring the SIPML5 and Asterisk Javascript library is required. We have everything installed and configured. The web softphone registers to Asterisk, sends and recei...
I'm looking for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom V...for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom VoIP libraries to make IVR calls. Here's a brief on what I need: - IVR Features: The bot should have interactive menu options and play text-to-speech messages. - SIP Server Integration: You'll need to integrate the bot with an existing SIP server. Ideal Freelancer: - Proficient in Python, especially in developing Telegram bots. - Experience with VoIP libraries, particularly in the context of IVR calls. - Familiar with SIP server integration. - Strong understanding of DTMF technology. If you're confident in your ability t...
...has a surprise face looking to the donkey but pointing to the kid with its trunk. - the third scene is the same as the first one, but the bear is flat on the floor after trying to grab the kid, but the kid avoided it jump and is about to fall on top. The bear some how expression suffering when hitting the floor. - the forth scene is similar to the first one, but now the kid is restrained by the elephant, the kid has fear in his face, the bear looks contempt and the hedhog and donkey looks with concern to the kid. - the fourth and last scene has the kid hanging from the claws from the bear unconscious, the hedhog and the donkey are looking with sadness the kid and the elephant is pointing to the kid with the trunk and looking to the donkey with its face saying something ...
Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.
We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...
...intensive research. Expertise working on softwares, call center, robocaller and conversational AI outbound calls for call center. The ideal candidate should have a proven track record in creating, managing or implementing cloud-based robo calling systems and also conversational AI calls that is good with asian language, especially Bahasa Indonesia. the subject of the research must have : telephony (SIP/VOIP) a working AI model or open to integrations with AI Basic Telco Services features such as, Campaign Calls, scheduler, call recorder) Wide system integration capability Macro/logic/script editing(for robo caller,pre recorder or AI generated TTS) white label ready or not. KPI analysis The Goal of this research is to figure which or what is the best system/platform/Integratio...
I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error. [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descr...
Creazione di un software che da un lato dialoghi tramite AMI (Asterisk management interface) con un centralino e dall'altro in base a delle condizioni impostabili, invii del messaggi tramite un gateway whatsappa già pronto chiamato che permette di inviare (pagando) messaggi illimitati tramite WhatsApp. inoltre questo software deve poter inviare dei messaggi anche da una lista di numerazioni che gli si fanno caricare.
I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descri...
We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...
...to 8pm Highcliffe Castle Rothesay Drive Highcliffe BH23 4LE Dorset Wine Festival Logo IMAGE 3: Mehroz Designer Jewellery Logo Sparkle Between Sips: Discover Mehroz Designer Jewellery at Dorset Wine Festival Join us on 5th & 6th October From 10am to 8pm Highcliffe Castle Rothesay Drive Highcliffe BH23 4LE Dorset Wine Festival Logo IMAGE 4: Mehroz Designer Jewellery Logo From Grape to Grace: Sip, Sparkle, and Shine with Mehroz Designer Jewellery at Dorset Wine Festival 5th & 6th October From 10am to 8pm Highcliffe Castle Rothesay Drive Highcliffe BH23 4LE Dorset Wine Festival Logo IMAGE 5: Mehroz Designer Jewellery Logo A Toast to Timeless Beauty: Discover Jewellery Aged to Perfection at Dorset Wine Festival 5th & 6th October From 10am to 8pm Highcliffe Castl...
Looking for a creative mind to help design and create an augmented reality video revolving around beauty and hair routines. Your mission, if chosen, is to captivate users with an interactive, virtual beauty experience. The scope includes: - Develop an AR video of 3d product over a trunk, the "fake" 3d products has to be made from 1 product picture . You must find a good video from adobe stock The ideal candidate: - Has solid experience in AR video creation - Has a keen understanding or background in beauty or skincare industry is a plus - Can understand and conceptualize my vision for the AR beauty experience Let’s talk!
As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.
I'm seeking a talented and creative graphic designer to help me envision and render a branded truck transformed into Optimus Prime from the Transformers franchise. This project involves both visual and mechanical transformation design. The primary skills required include: - Expertise in logo and print design and web design - Familiarity with the Transformers series, specifically the character of Optimus Prime - Understanding of machinery and vehicle structure Your responsibility will be to: - Create concepts for how we can visually morph our existing truck design into a representation of Optimus Prime - Give inputs on possible mechanical modifications that would complement the visual transformation - Prepare a visually appealing presentation of the design c...
We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...
I'm looking for a seasoned app developer with expertise in Linphone and SIP to help with customizing the open-source VoIP service. In this project, you will: - Compile Linphone with my brand logo, offering a personalized look to the application. - Integrate the app with my push server. This will involve modification of existing features to ensure a seamless flow between the app and the server. Given that the specific features to be modified weren't mentioned, I assume that you will provide the details later on or the freelancer will need to assess the necessary changes upon reviewing the current status of the app. Proficiency in VoIP/SIP and experience in both Android and Apple app development is required. Familiarity with app store submission procedures will be...
I require a designer to bring my l...to make it modular. - Create scenes of the robot in agricultural terrains: example of scenes: 1- The robot is on a ramp trying to pull baskets of cultivated fruits 2- Multiple robots on a field trying to move rocks or parts of trees, others cultivating, 3- Robots cultivating in vineyards 4- Robots working in a stormy weather 5- The robot driving an agricultural car while other robots are in the trunk cutting tree branches Key Skills Required: - Proficiency in a 3d design software. - Prior experience in product (robot) designs is ideal. - Understanding of industrial-use requirements for robots. As the purpose of this robot is industrially oriented, a level of experience in designing industrial products and robots would be a significant advantag...
I'm seeking an expert in SIP and VoIP to assist me with various issues within my PortaOne system. Key Requirements: - Extensive knowledge of SIP and VoIP solutions, particularly with PortaOne - Proficiency in call routing and configuration for diverse scenarios - Proven experience in troubleshooting and resolving various issues - Clear communication skills with the ability to provide detailed explanations and solutions.
We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...
I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50
We are looking for an engineer proficient with Raspberry Pi, as we are in need of developing a VoIP PBX system on a Raspberry Pi 3 Model B+ . The end goal includes the integration of specific features into the system such as: - Call Recording - Voicemail to Email - Conference Bridging - Additional bespoke features Your expertise should include not only Raspberry Pi but also Asterisk and RASPBX/FreePBX. We are aiming for a robust, stable, and user-friendly system with custom features tailored primarily to business needs. The successful contractor will be required to develop the system on his own Raspberry Pi and submit an IMG file for loading onto other Raspberry Pis. The successful contractor will be working with our software developers so bids from individual contractors only w...
I require a designer to bring my ...make it modular. - Create scenes of the robot in agricultural terrains: example of scenes: 1- The robot is on a ramp trying to pull baskets of cultivated fruits 2- Multiple robots on a field trying to move rocks or parts of trees, others cultivating, 3- Robots cultivating in vineyards 4- Robots working in a stormy weather 5- The robot driving an agricultural car while other robots are in the trunk cutting tree branches Key Skills Required: - Proficiency in a 3d design software. - Prior experience in product (robot) designs is ideal. - Understanding of industrial-use requirements for robots. As the purpose of this robot is industrially oriented, a level of experience in designing industrial products and robots would be a significant advantage...
...Jose do Rio Pardo, this coffee reflects a tradition of excellence since 2003. It's part of the Mata Atlantica project, underscoring a dedication to rainforest conservation. Indulge in a cup that celebrates both exceptional taste and ecological commitment. For the 6th image which is the Colombia photo of a street in Cartagena there's also quite a bit of text to add: Savour the luxury with every sip of our Colombian House Blend, a symphony of caramel, chocolate, and a hint of orange zest. This exquisite blend is a testament to the lush landscapes at the Andes' foothills, where our Colombian Excelso beans flourish in the nutrient-rich volcanic soil, capturing its essence to perfection. Carefully roasted to a medium perfection, our Excelso beans unveil a delicate b...
Necesito ayuda con la configuración de mi Asterisk, ya que no soy capaz de recibir el DTMF, con la opción que selecciona el usuario, tras escuchar la locución de la IVR. He intentado transcodificar la señal, y aplicar diferentes configuraciones, y codecs sin éxito.
looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related applications
...tithe underneath it but the label should only have english words on it. This logo will likely be redone over but if you include a new logo in your design that I like I will pay extra for it but the main focus is the label. The label size is 19cm x 15cm The bag is white. I need the front and back label. ~~~ Front label text: Sharpen Focus Boost Energy Strengthen Immunity Calm Stress Every Sip Supports French Vanilla 100% Organic Non GMO 30 Day Supply Net Wt 12.7oz (360G) -------------------------------------------- ~~~ Back label text: The nutritonal facts are attached. Store in a cool, dry place to preserve freshness. Brew the Perfect Cup For the ideal experience, use 1 tablespoon of coffee for every cup (8 oz) of water. Adjust to your taste. Our Mission At '...
I'm in need of a proficient specialist to establish my SIP trunk and an IPPBX server. Here are the functionalities I require: • Making both inbound and outbound VoIP calls. • Call recording, call forwarding, and voicemail to email features for my IPPBX server. You'll also need to set up three or more SIP trunks. Experience with VoIP technologies and SIP systems is crucial for the success of this project. Knowledge of REST APIs will be advantageous. Please ensure your proposal demonstrates your related past projects and expertise.
We are a vibrant cold-pressed juice bar on the lookout for an imaginative and skilled graphic designer to revamp our digital screen menus and create a detailed standing printed menu. Our brand prides itself on delivering health through every sip, offering a range of cold-pressed juices, smoothies, and health shots. We aim to capture the essence of our healthful and flavorful offerings in our menu designs, adhering to our minimalist brand aesthetic. Objectives: Digital Screen Menus: Design three clear, captivating digital screen menus that showcase our product range with prices. The design should be easily readable from a distance and engage customers with its visual appeal while following our minimalist brand guidelines. Printed Menu Design: Create a detailed printed standing menu...
We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...
I am urgently seeking an experienced telephony and data processing specialist to configure my Grandstream UCM6302A with Asterisk. The core functionality required includes receiving calls, playing a welcome message, meanwhile working with Caller ID and Web API to determine where to forward the call. When a call comes in, • first a welcome message is played () • in the meantime the caller ID will be sent to web API preferably POST, but get can be if POST is not possible () •The API will respond json array: - {forward_to: 33356853 } - Forward the call to 33356853 - {forward_to:0 } - Play message and terminate the call • If forwarded call is not answered by Agent in three rings, another call to API will
...am looking to incorporate AI features into my call center system, specifically Vicidial and Asterisk. As these platforms form the core of our operation, it is essential that any alterations enhance our outlay without disrupting the existing structure. Key Aspects of the Project: - AI Implementation: Even though I haven't specified the exact AI features to integrate, I'm interested in potential focus areas such as speech recognition and transcription, natural language processing, or sentiment analysis. Proposals that offer comprehensive strategies addressing these or other AI fields will be highly considered. - Dual Integration: The AI features must be incorporated into both Vicidial and Asterisk, aligning and harmonizing their performances. - Efficiency Goal: ...
i need someone to teach me how to upgrade firmware of cisco 7821-k9 to make it use sip protocol to hook it up on asterisk pbx More details: Which specific features do you require for your Cisco 7821-K9 SIP protocol project? upgrade firmware , connect it to asterisk as sip extension Which version of firmware would you like to upgrade your Cisco 7821-K9 SIP protocol to? Latest firmware version What functions do you require for the SIP extension with your Asterisk system? Call Recording, Call Transfer, Multi-Line Functionality thank you very much
I require the expertise of a skilled freelancer to assist in the configuration of our Sangoma Vega 400g Voice Gateway device. It's critical that you have familiarity with E1 ISDN to SIP trunk conversion, due to our primary objective being system compatibility upgrading. Key responsibilities: - Customize settings on Sangoma Vega 400g Voice Gateway according to specific requirements. - Setting up a connection with Swedish Telco, Telia. Ideal Candidate: - Previous experience with Sangoma Vega Gateway configuration. - Knowledgeable in E1 ISDN to SIP trunk conversions. - Familiar with Telia's connectivity. - Quick and efficient problem solver, without sacrificing quality results.
I am looking for a specialist to integrate WhatsApp with SIP on Linux, focusing primarily on enabling audio communication and text messaging functionalities. The end goal is to create a seamless communication platform for efficient interaction. Key tasks include: - Designing and implementing a solution that allows WhatsApp to work in sync with SIP protocol - Ensuring the integration supports both audio calls and text messages. A deep understanding of both WhatsApp API and SIP protocol is essential, as well as experience in Linux system administration. Knowledge of VoIP would be a plus. Your delivery should be a working, robust integration that ensures smooth and seamless operative communication. Please bid with your proposed solution.
I have a Twilio account with sip trunking set up, and I've install Asterisk on Arch Linux, I've attempted to set up the config but have not been able to. I'm looking for someone to set up a basic config where I can send and receive phone calls. The details don't matter, I just want to get it working so I can adjust it once the simplest config is working. If interested please bid the amount you are able to do this for, and include the word "briefcase" in your bid so I know you've read the description and can complete the project for the amount bid.
se necesita un software que permita gestionar las llamadas voip configurable de preferencia que admita SIP Trunking, el aplicativo debe permitir gestionar las llamadas, medir el rendimiento del usuario que realiza las llamadas, debe tener tener un marcador de numero al que se va a llamar configurable(auomatico / manual / random) de preferencia el freelance debe tener un proyecto similar o terminado
I'm seeking a highly competent Investment Advisor who can guide me in the areas of Insurance, SIP and Policy. The role is focused on life insurance advice and strategies for robust legacy planning. - Expertise: You should have extensive experience and knowledge of life insurance products and policies, Systematic Investment Plan (SIP), Insurance policies. - Task: I require in-depth analysis of my current situation and guidance on where to direct my resources for optimal legacy planning. This will include assessing risks, evaluating potential returns, and designing a balanced legacy-focused investment plan. - Aim: Primarily, you will help me plan for risk cover and wealth accumulation, while particularly focusing on legacy planning via life insurance products. Plea...
...featured communication app for both iOS and Android. This app will connect with my existing Asterisk server through APIs. Key Features Include: - User creation - Real-time balance display - Call-making functionality - Fully integrated payment gateway - Text messaging - SIP voice calls (Not video calls, just normal SIP calls) Necessary Skills and Experience: - Proficient in iOS and Android app development - Proven experience with PortSIP SDK - Familiarity with Asterisk and its relevant APIs - Skills in developing chat features, specifically text messaging and voice calls, within an app - Experience in implementing a payment gateway in an app Please note, I have access to and can provide the necessary Asterisk API documentation. Ideally, you are able ...
I need assistance setting up a new SIP Trunk on my FreePBX 14 system. Here's what you will need: • Experience with FreePBX 14 • Familiarity with the setup of SIP Trunks • Knowledge of goip local voice gateway Tasks to be accomplished: • Set up a new SIP Trunk on my FreePBX 14 • Ensure the setup is correctly done using my service provider, goip local voice gateway. Only those with the prescribed skills and experience should bid, as I need this done quickly and accurately. Thanks!
...application for both Android and iOS platforms that focuses on SIP (Session Initiation Protocol) registration. This will directly influence the call functionality, effectively enabling calls to come from a mobile sim and not from a server. Key features: - SIP registration: This is a vital requirement. The application needs to have the capability to register with the SIP I will provide. - Voice-activated interface: User preference is for a voice-activated interface. Building a user-friendly interface that reacts to the voice command, "Register SIP" is essential. Ideal skills and experiences: - Experts in mobile application development, particularly in Android and iOS. - Proficient in building voice-activated interfaces. - Familiarity with SIP...
...successfully create code that is capable of making a call to a specific number utilizing a SIP user and play the audio file generated using a file generated by combining TTS output and static audio . The project has specific requirements, including: - The use of a Python programming language, due to its efficient and fast execution. - An essential feature includes utilizing a free Text to Speech (TTS) system, in order to relay a pre-determined message. - Additionally, I would like the implementation of static audio that can be mixed in with the TTS function. - it will be good if it can be diverted to agent if user enter some specific DTMF tone I have a cloud vcdial account and i assume the user account are SIP. I can give access to the person remotely so he can verify t...
I need a skilled developer to build a mobile SIP registrar application for both iOS and Android platforms. This application's primary purposes are making and receiving voice calls and managing SIP accounts, so proficiency in VoIP technology and SIP protocols is a must. Not only should the application be functional but it should also have a modern and stylish user interface. Therefore, a candidate with a keen eye for design is preferred. However, the functionality remains the top priority over visuals. If you have experience in this type of software development along with a knack for creating user-friendly apps with a modern touch, we should talk.
Hello, I operate a fax communication system leveraging Hylafax, integrated with an Asterisk server and iaxmodem, all running on Alpine Docker. While our outgoing fax functionality performs flawlessly, we are encountering persistent issues with incoming faxes. Specifically, incoming fax pages frequently get cut off midway, resulting in incomplete document reception. We are in search of a seasoned Hylafax professional who can diagnose and rectify this particular issue. Expertise in managing Alpine Docker environments and Asterisk/iaxmodem configurations will be highly regarded. Desired Expertise: Demonstrable experience with Hylafax, especially in fixing issues related to incoming faxes. Deep knowledge of Asterisk and iaxmodem. Proficiency with Docker containers, pre...
