Sip trunking asterisk avaya työt

My SIP Trunking system is constantly preventing me from receiving inbound calls. It's a constant issue that requires urgent attention. Ideal candidates will possess: - Extensive experience with SIP Trunking - Knowledge in diagnosing and resolving call reception problems - Proficiency in troubleshooting technical issues constantly, not sporadically.

Estamos à procura de um especialista em Issabel e Asterisk para implementar um sistema de Resposta Vocal Interativa (IVR) na nossa plataforma de servidor Issabel 4.0.0-6. O objetivo deste sistema é realizar entrevistas telefónicas automatizadas onde os respondentes possam responder a perguntas pré-gravadas utilizando o teclado do seu telefone. O sistema deve ser capaz de carregar uma base de contactos para realizar automaticamente as chamadas telefónicas e capturar estas respostas numa base de dados para análise subsequente. Além disso, o sistema IVR deve incluir a funcionalidade de conversão de texto em voz (TTS) para facilitar a criação e atualização de prompts de voz. No entanto, também d...

I am looking for a competent professional with experience in Freepbx and Avaya IP integration. The primary objectives of the project are call routing, call forwarding, and establishing a unified voicemail. What I am looking for: - Experience with Freepbx and Avaya IP office 500 V2 systems - Proven track record with telecommunication technology - Capability to integrate functionalities even with gaps in specific call routing or forwarding options Please note that while some specifics on the call routing and forwarding options are not outlined, I expect freelancers to provide suggestions based on their experience and expertise. I look forward to working with an individual who can make these systems work smoothly together.

...solution, I'm seeking guidance from a skilled freelancer experienced with both Raspberry Pi and Asterisk server setup. Key Project Details: - I do not require a full-fledged PBX setup, just a basic VoIP service facilitated through my Raspberry Pi. - The primary feature I'm interested in implementing is voice calling. Required Skills: - Proficiency in configuring an Asterisk server on Raspberry Pi. - Strong understanding of VoIP and related protocols. - Ability to guide and explain the setup process clearly. Your Role: I've tried to set it up but no voice can be heard on the other end. So this task is mainly a trouble shooting job. Your primary role will be to walk me through the setup of Asterisk on my Raspberry Pi, ensuring proper configuration f...

I'm in need of a highly qualified and certified network and security engineer. Ideally, you should have certifications from Cisco, F5, Symantec, HP, HPE, Dell, Huawei, Avaya, Fortinet, and Juniper. Key Responsibilities and Tasks: - Network design and configuration - Firewall management - Network troubleshooting - Telecommunication system setup - Network security audits - Hardware installation and maintenance - System integration - Vendor selection and management - Providing ongoing support Your background should demonstrate extensive experience with all the above functions. I'm particularly interested in freelancers who have successfully handled similar tasks in the past. Your success will hinge on your ability to improve our overall network and security infrastructure. ...

I require an expert in Asterisk and FreeSWITCH development, along with experienced system engineering skills. Specifically, I need help with: - Asterisk and FreeSWITCH development - Integrating VoIP - Setting up Call routing and IVR (Interactive Voice Response) - CTI - Skill group base call routing to agents. Apart from these, the implementation of Asterisk and FreeSWITCH clustering as well as Call Center Reporting is required. The technology stack should be Linux, PHP, MySQL along with postgresql and API knowledge. The freelancer should have considerable experience in these to meet the project's specific requirements.

I'm looking for an experienced freelancer to help me set up the SIP locally to connect incoming calls to an AI script I have. This setup has to be done on VICIdial using the QuestBlue SIP. The main objective of this project is to establish a call response system. As someone with a pre-built AI script, I need you to link it to the incoming calls on the VICIdial platform. Your expertise will include, but not be limited to: - Proficiency in VICIdial: You must have previous experience working with this platform. - Familiarity with SIP set-ups: QuestBlue SIP experience would also be a bonus. - AI Integration: Able to connect incoming calls to an AI script. Please ensure these skills align with your capabilities before bidding on this project.

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...

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...

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.

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'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...

...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...

...seeking a VoIP consultant for improvement of my existing computer-based VoIP system. The purpose of the project is twofold - improved communication efficiency and enhanced call quality. Key Tasks: - Analyzing the current computer-based system setup - Implementing the connection of Physic SIP to asterisk on the cloud for enhanced call quality Ideal Skills and Experience: - Proven experience as a VoIP consultant - Excellent knowledge of IP PBX system - Experience with connecting Physic SIP to asterisk on the cloud - Ability to improve communication efficiency and call quality. Kindly submit your proposal outlining your plan to achieve these two goals along with your previous relevant work. Looking forward to finding a VoIP specialist who can provide a swi...

I'm looking for a skilled artist with expertise in Pop Art to conduct a 2-hour painting and drawing workshop at a "Paint and Sip" event for National Women's Month. Your main task will be to teach college students, who are beginners in the art world, how to create their own pieces. Here's what you need to bring to the table: - Proven experience in Pop Art - Prior teaching experience, particularly with beginners - An approachable and patient attitude, encouraging students step-by-step - Capability to manage a large group and ensure everyone understands the process Offer students an opportunity to learn something new and have fun while doing it. Apply if you are passionate about art and enjoy sharing your knowledge.

I am looking for an expert in Yeastar p560 VoIP systems to implement a solution for a trunk to trunk call forwarding inclusive of Direct Inward Dialing (DIDs). The task is trou...forwarding inclusive of Direct Inward Dialing (DIDs). The task is troubleshooting headers, the trunks / solution is already setup. The key functionality includes: - Trunk to trunk hand off with headers (in and out) - Forwarding calls over VoIP Your expertise should span the configuration of VoIP systems for diverse and specific users. Knowledge of wireshark and pcap analysis, understanding of SIP headers. With your understanding of my needs, you'll enable seamless communication within my organization. Prior experience in similar setups is a necessity. Your bid will be a value-add to the success o...

I am seeking an experienced developer to create an application that will be compatible with both iOS and Android platforms. The primary function of this application will be voice calls. Key Features: - SIP registration: You will be provided with the SIP account username, password, and IP from my soft-switch to integrate with the app - Ability to make calls from mobile SIM, not server - An integrated contact list The ideal candidate will have experience in developing voice call applications and have a good understanding of Session Initiation Protocol (SIP). Knowledge and experience with mobile application development for both iOS and Android platforms are essential. Your expertise in integrating apps with contact lists will be particularly valuable for this project.

As the project owner, I am looking for a seasoned Flutter developer adept in creating applications for both Android and iOS platforms. My project primarily involves: 1. SIP Account Registration: You'll be required to implement a feature that allows users to easily register a SIP account. 2. Call Functionality: You will be in charge of creating a phone call feature that operates using a SIP server. 3. Simple User-Interface: The app should have a user-friendly interface, aiming for simplicity and easy navigation. The ideal candidate should have proven experience working with Flutter and SIP technology, and should be proficient in developing applications for both Android and iOS.