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    We are in search of an experienced freelancer, skilled in Bitrix24 Rest API and capable of system integra...in search of an experienced freelancer, skilled in Bitrix24 Rest API and capable of system integration. Tasks Included: * Integration of our Bitrix24 with Cisco Unified Communications Manager. Experience in these systems is a must. Skills & Experience: * Deep understanding of Bitrix24 and its Rest API. * Proven experience in Cisco Unified Communications Manager. * Dexterity in the development of custom API endpoints. * Proven ability in Bitrix24 module customization. * Background in system integration. Your responsibility will be to create a seamless and effective integration between our Bitrix24 system and Cisco UCM. Apply if you confid...

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    ...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...

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    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. ...

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    I am looking to set up an Enterprise MFA VPN solution. The freelancer best suited for this role would ideally have: - Proficient knowledge in cloud and on-premises VPN setup - Experience implementing MFA VPN - Cisco ISE Authorization(Posture policies).

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    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.

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    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.

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    I currently have enhanced licenses for Cisco CUCM 11.5 and I'm seeking to upgrade to the Flex system within a month. While I've skipped explaining the specific functionalities I'm looking to add with the Flex system, ideally, the prospective freelancer should be well-experienced in transitioning from Cisco CUCM's enhanced licenses to Flex. Desired Skills and Experience: - Proficiency in Cisco CUCM 11.5 licensing and operations - Experience in upgrading to Flex system - Strong understanding of Flex system functionality options - Ability to recommend suite of functionalities to maximize collaboration, mobility and contact center features. - Ability to deliver the project within a month time frame.

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    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...

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    ...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...

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    For your quote to be considered please start with "I am a Cisco Professional" I'm looking for a professional who can assist me in installing Windows Server OS on my Cisco UCS C220 server as soon as possible. We have the server and have configured the LSI Virtual Drives when we try to install the OS we get Driver IRQL error with "What Failed " For someone who is experiences this should be a quick one

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    I'm in need of a...help install and set up an Alcatel PBX system for a medium-sized enterprise. The project also involves integrating this PBX system with VoIP services. This will be remote project, our team will give you access from the site. Key Responsibilities: - Configuration of Alcatel PBX system OXO408 - Integration of the PBX system with VoIP services along with call recording. - Integration with existing Cisco IP Telephony Ideal skills for this job include: - Profound understanding of Alcatel PBX systems. - Experience with VoIP services integration. - Prior work on medium-sized enterprise deployments. Your expertise in these areas will help us to establish a reliable communication system for our business, ensuring seamless connection with our clients and within ou...

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    Python Telegram Bot w/ VoIP 1 day left
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    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...

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    SRP Consulting -- 6 1 hour left
    VERIFIED

    We are looking for c...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...

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    ...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...

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    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...

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    experienced network specialist, well-versed in Cisco ISE and firewall configurations for an enterprise level network. We're currently experiencing network connectivity issues and require assistance with firewall configurations and overall network security.

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    Cisco Packet Tracer make the activity lab

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    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...

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    I'm searching for an expert to help me set up a honeypot in Cisco Modelling Labs (v2.6.0). The primary purpose is to monitor network traffic, identify, and capture any malicious activities that happen within the network. It can be a low or low-medium interaction honeypot. As a beginner, I require an individual with skills and experience to navigate these systems. Mastery of Linux server for implementing the honeypot is essential due to its build on a Linux system. Need it completed before the end of April. Key Competencies: - Expertise in Cisco Modelling Labs - Proficiency in with Linux - Understanding of network monitoring and security principles - Ability to train a beginner on how to run the honeypot with documentation (pictures) of how the honeypot was created.

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    We are looking for c...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...

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    Hi, I have one sample codes which work as voip client . it should run on ARM32 . and I have one rockchip rv1106 linux SBC (I can add memory and emmc size following your advise). this rv1106 has its SDK, I installed on virtual unbuntu 20.04,and buid SDK and its sample application codes ok, but when I integrate the voip sample code into sample application codes ,it has link error about GLIBC,etc ,similar outputs as this url: I tried to fix it myself ,and asked for help ,still not working .I guess rv1106 cross compiler tool shipped by chip provider is uclib (arm-rockchip830-linux-uclibcgnueabihf ) ,but the voip sample codes need glib. so I am thinking maybe we should use buildroot to build a new linux image

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    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.

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    Study the scenario NOTE: You will use this scenario to answer the rest of the project. You are supposed to use Cisco Packet Tracer in this project. Belmand College Belmand College is a registered private higher education institution in South Africa. It specializes in training IT Diplomas and Professional Courses with its Campus located in Midrand, Midrand Campus has 3 buildings which houses the different departments, lecture rooms and labs. Building A houses the IT Department, Library and Main Administration, the other buildings B and C houses lecture rooms and labs. The college has a total enrollment of about 2000 students. They have over 100 employees which includes admin employees, full-time lecturers and part time lecturers are hired every semester. They are currently u...

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    We are looking for c...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...

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    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...

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    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.

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    I need a technical consultation for current project to migrate UCS Manager to Cisco Intersite. Please be someone with a expertise level.

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    We are looking for c...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...

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    I'm seeking a talented Node.js coder to enhance my existing Mediasoup-based video conferencing tool. The primary purpose of this project is to include an RTP recording feature. Here's what I require: 1. **Execution of RTP recording:** The virtual rooms' audio and video should be captured and saved. Any experience with similar projects ensures you understand the specifics. 2. **Storage Location:** The recorded files must be saved on our local server storage. Knowledge of server management and storage distribution will be necessary to ensure efficient storage. 3. **File Format:** The recorded files will need to be saved in the WEBM format to guarantee consistency and usability. 4. **File Structure:** The video and audio from every conference must be compil...

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    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

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    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...

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    I urgently need an experienced Cisco Webex Programmer that can handle a specific task. The task involves writing a script, as well as establishing an xapi event for the UI. This should result in switching the inputs on the Cisco Webex Dual 70. This change should be triggered by user interaction. This is a task for an experienced programmer only, ideally with a strong history of Cisco Webex custom integration. Those with proficiency in user access control and meeting scheduling and management are highly sought. The job will require expert problem-solving and technical skills. Main Tasks: - Write a coded script - Implement an xapi event for the UI Trigger Condition: User interaction Prior Expertise: - Cisco Webex programming - User access control - Meeting sc...

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    ...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...

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    Hi everyone, I will need a computer network analyst (ideally with cisco certification) with experience in measuring network delays, calculate jitter (with tools povided), do some graphs in excel etc. Analytical specific requirements in chat Deadline: 4/4 Budget: $20 USD

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    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

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    I am currently looking for a proficient CCNA lab trainer who is capable of providing a range of assistance. Key Responsibilities: - Setting up and configuring the lab - Troubleshooting t...providing a range of assistance. Key Responsibilities: - Setting up and configuring the lab - Troubleshooting throughout the process - Network simulation to ensure optimum functionality The selected freelancer should be comfortable with both onsite and remote training, thus ensuring maximum flexibility. We will be working with a variety of Cisco devices which include Routers, Switches, and Firewalls. Ideal Skills: - Excellent knowledge in working with Cisco devices - Prior experience in CCNA lab training - Proficiency in Network simulation - Strong troubleshooting and networking con...

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    ...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 '...

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    Hi everyone, I will need a computer network analyst (idially with cisco certification) with experience in packet tracer software, to do a simple network, calculate jitter (with tools povided), do some graphs in excel, calculate some delays etc. Analytical specific requirements in chat Deadline: 4/4 Budget: $60 USD

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    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.

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    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...

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    ...BIM/MEP Engineer, proficient in utilizing Revit, AutoCAD, and Navisworks. The selected candidate's scope of tasks will encompass: - Assisting with comprehensive 3D modeling - Conducting effective clash detection - Providing technical support - MV/LV Networks , (cable routing, medium voltage & low voltage switch gears, transformer loading). - load calculation and load estimation according to SBC code - power single line diagram cable sizing, voltage drop, and short circuit calculation. Sockets, power outlets. - Load schedule. - street and exterior lighting / Interior Lighting design. - Grounding systems. - ICT - Sound system, - Cctv system, Ideally, the freelancer should have experience not only in design but in design implementation as well. A proven track record o...

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    This is a task for school assignment. You will implement a streaming video server and client that communicate using the Real-Time Streaming Protocol (RTSP) and send data using the Real-time Transfer Protocol (RTP). Your task is to implement the RTSP protocol in the client and implement the RTP packetization in the server. The doc has explanation in detail. The code is already half done but need to fix it. Will be sharing the code Timeline - 4 Hours Requisites- Proficient in Java & has used RTSP protocol before.

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    We are looking for c...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...

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    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

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    ...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: ...

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    I require a professional who is adept with Cisco's hardware. CISCO Wireless setup with C3504 controller and ACCESS Points 2800 series there in existing network . Need to add 8 APs 3000 series to this network . Remote login will be provided.

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    I'm in need of a Cisco engineer with experience in setting up network infrastructure. The project includes: - Setting up infrastructure for a medium-sized business - Primarily involved in routers and switches setup The ideal candidate should have proven skills and experience in network engineering particularly in setting up routers and switches. A Cisco certification would be an added advantage.

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    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

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    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.

    $176 (Avg Bid)
    $176 Avg Bid
    5 bids