Filtrér

Mine seneste søgninger
Filtrer ved:
Budget
til
til
til
Slags
Færdigheder
Sprog
    Job-status
    2,000 sip asterisk jobs fundet, i prisklassen EUR

    Integrate Asterisk + UniMRCP + AWS LEX

    €171 (Avg Bid)
    €171 Gns Bud
    1 bud

    I m working in voip in Morocco from 2014 I have a system of asterisk Isabel the problem is when my client send me trafic give him an erreur 486 must be 503 In this case my client lost his traffic https://www.freelancer.com/users/l.php?url=https:%2F%%2Fusers%%3Furl%3Dhttp:%252F%%252Fpt%252FA42hBUju%26sig%3D50c67e1b3a7f27f20771fd2ee320f07b9eb302904f9e19edea3a28d328aafbdf&sig=dbffe148af4b5367f232a97a6acda81aea4364dfabcb09ccf07b413f3f1c3be9

    €386 (Avg Bid)
    €386 Gns Bud
    6 bud

    We must have set up a new asterisk server that follows the normal asterisk guidelines. We also need a script where we provide the telephone number (DID), opening hours also agents who are queuing, then it must do the setup in configs.

    €166 (Avg Bid)
    €166 Gns Bud
    4 bud

    Asterisk Mikrotik

    €2 / hr (Avg Bid)
    €2 / hr Gns Bud
    1 bud

    HI Asterisk ?

    €14 (Avg Bid)
    €14 Gns Bud
    1 bud

    asterisk,php,ivr,dtmf,TTS,ASR,php

    €1127 (Avg Bid)
    €1127 Gns Bud
    1 bud
    €185 Gns Bud
    1 bud
    Asterisk Job Udløbet left

    Asterisk Job

    €278 (Avg Bid)
    €278 Gns Bud
    1 bud
    For AsteriskKingx Udløbet left

    Db Fix for Asterisk King , As discussed

    €9 - €51
    €9 - €51
    0 bud
    Asterisk discussion Udløbet left

    Asterisk discussion

    €9 (Avg Bid)
    €9 Gns Bud
    1 bud

    Vb6 ivr asterisk dialogic programme

    €28 / hr (Avg Bid)
    €28 / hr Gns Bud
    1 bud

    Sonetel & Asterisk Configuration

    €20 / hr (Avg Bid)
    €20 / hr Gns Bud
    2 bud

    a dialer for my asterisk box kasdjlasjdlasjdlasjdlasjdalsdjalsdjalskjdalskjdalskjdalksjdalskjdalskjdlksajdlasjdalsjdalsjdalsjdalsjalskjd 100chars

    PHP
    min €33 / hr
    min €33 / hr
    0 bud
    asterisk dialer Udløbet left

    a dialer for my asterisk box kasdjlasjdlasjdlasjdlasjdalsdjalsdjalskjdalskjdalskjdalksjdalskjdalskjdlksajdlasjdalsjdalsjdalsjdalsjalskjd 100chars

    PHP
    €140 / hr (Avg Bid)
    €140 / hr Gns Bud
    1 bud
    Asterisk+Zendesk Udløbet left

    Добрий день, Мені потрібно виконати інтеграцію Asterisk+Zendesk - бажано, щоб при надходженні дзвінку створювався тікет, ідентифікувався дозвонювач, і т.д. Чи є в Вас такий досвід роботи? Ми знаходимося у Києві. Бюджет та подробиці оговоримо.

    €9 (Avg Bid)
    €9 Gns Bud
    1 bud
    SRP Consulting -- 2 6 dage left
    VERIFICERET

    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...

    €287 (Avg Bid)
    €287 Gns Bud
    3 bud
    ERPnext with Twilio 6 dage left
    VERIFICERET

    ...Twilio integration for incoming and outgoing voice calls. The productive system should be hosted on a Digital Ocean Droplet. A development system is also needed. Both production and development should be encapsulated in Docker containers. Twillio: I am a specialist in the sales of IT services. What I need are direct dialing, fax reception (no joke, German authorities) and call forwarding or sip trunking to my office service. I mostly make phone calls from Croatia to Germany, Austria and Switzerland. In addition, telephony should be integrated into ERPnext. An important point is call handling. If someone dials 0 (central office), they should be forwarded directly to the office service. The same thing should happen when I'm on the phone and someone calls my extension....

    €2225 (Avg Bid)
    €2225 Gns Bud
    11 bud

    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

    €102 (Avg Bid)
    €102 Gns Bud
    25 bud

    ...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: ...

    €451 (Avg Bid)
    €451 Gns Bud
    24 bud

    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

    €36 (Avg Bid)
    €36 Gns Bud
    7 bud

    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.

    €164 (Avg Bid)
    €164 Gns Bud
    5 bud
    WhatsApp-SIP Integration 4 dage left
    VERIFICERET

    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.

    €272 (Avg Bid)
    €272 Gns Bud
    16 bud

    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.

    €56 (Avg Bid)
    €56 Gns Bud
    11 bud

    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

    €158 (Avg Bid)
    €158 Gns Bud
    14 bud

    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...

    €263 (Avg Bid)
    €263 Gns Bud
    2 bud

    ...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 ...

    €474 (Avg Bid)
    €474 Gns Bud
    44 bud

    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!

    €216 (Avg Bid)
    €216 Gns Bud
    8 bud

    ...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...

    €206 (Avg Bid)
    €206 Gns Bud
    38 bud

    ...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...

    €99 (Avg Bid)
    €99 Gns Bud
    9 bud

    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.

    €180 (Avg Bid)
    €180 Gns Bud
    25 bud

    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...

    €26 / hr (Avg Bid)
    €26 / hr Gns Bud
    14 bud

    ...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...

    €11 (Avg Bid)
    €11 Gns Bud
    3 bud

    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.

    €232 - €695
    Lokal
    €232 - €695
    0 bud

    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...

    €89 (Avg Bid)
    Haster
    €89 Gns Bud
    10 bud

    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.

    €181 (Avg Bid)
    €181 Gns Bud
    8 bud

    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.

    €124 (Avg Bid)
    €124 Gns Bud
    32 bud

    I'm in urgent need of an individual adept in Voipfone SIP services and Java. Unfortunately, I don't have access to the Voipfone documentation and API credentials. Hence, a freelancer who is already versed in said provider would be preferred. The ultimate goal is to create a compact Java project that fetches caller ID information from Voipfone's SIP services. Key responsibilities: - Creating Java code for Voipfone SIP caller ID queries - Overcoming lack of direct API access Desired skills and experience: - Expertise in working with Voipfone services - Proficient in SIP technology - Strong knowledge in Java - Ability to work around API limitations effectively The project time frame is ASAP. So a freelancer capable of prompt and efficient service...

    €503 (Avg Bid)
    €503 Gns Bud
    42 bud

    ...needed marked with blue. Message on the back of “OLIVEIRA" ENGLISH: Enjoy the distinctive ‘Oliveira’ a single-origin coffee by Uai Coffee, crafted with care by the award-winning José Maria de Oliveira at the lush ‘Toca da Raposa’ farm, nestled at an elevation of 1,200 meters. This exceptional coffee, boasting notes of brown sugar, citrus, and orange, offers a unique sensory journey with each sip. Proudly embracing sustainable practices, José Maria’s dedication to quality shines in every cup. Discover more about our commitment to excellence and the stories behind our brews. Experience ‘Oliveira’—where tradition, sustainability, and passion for coffee converge. FRENCH: Savourez l'unique ‘Olivei...

    €21 (Avg Bid)
    €21 Gns Bud
    1 bud

    ...developer experienced in WebSocket/AudioSocket technologies and Asterisk integration to develop a solution that enables real-time transcoding with OpenAI Whisper through gRPC. Requirements: 1) WebSocket/AudioSocket Integration: Develop WebSocket/AudioSocket functionality to facilitate real-time audio communication with OpenAI Whisper. 2) gRPC Compatibility: Implement gRPC to ensure compatibility for seamless communication between components. 3)Real-time Transcoding: Enable real-time transcoding capabilities to convert audio data appropriately for interaction with OpenAI Whisper. 4)Asterisk Integration: Integrate the solution with Asterisk to allow seamless initiation and handling of audio calls from Asterisk dialplans. Example Asterisk Dialplan: [a...

    €159 (Avg Bid)
    €159 Gns Bud
    16 bud

    Task details - For ISUP/SS7 (E1 card implementation) along with IVR knowledge in any open source SIP servers like Freeswitch/Yate/Mobicents etc. or working experience in CRBT server. Interested candidates please apply

    €138 (Avg Bid)
    €138 Gns Bud
    3 bud

    I am searching for a skilled software developer with a strong background in Asterisk, Dialer, IVR and VOIP technologies. Although I haven't specified particular functionalities, general familiarity with call routing, call recording and interactive voice response (IVR) would be beneficial. The ideal candidate for this job should be proficient in: - Designing, implementing, and maintaining Asterisk software - Developing dialer functionalities, with emphasis on auto dialing, click-to-dial, and predictive dialing - Ensuring system is up-to-date and secure Freelancers who apply should provide any past work, detailing their experience and including project proposals, if any. If you believe that you have the expertise to effectively take on this project, I encourage...

    €6869 (Avg Bid)
    €6869 Gns Bud
    10 bud
    SRP Consulting Udløbet left

    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...

    €278 (Avg Bid)
    €278 Gns Bud
    1 bud

    We need to create an Asterisk aplication (v18) for Service at workshop by appointment for vehicles. This aplication must have voice recognition in English /Spanish language and Text to speach language with Google technology. Functionality: i) Welcome. ii) Select Languague. iii) Request Data: * Type of vehicle * City * Car licence plate * Telephone number. * Date request. * Time request. d) System will confirm first date/time available and customer will confirm. At this time, application will not have conectivity with real system....only must confirm next day and time users told. But it will have errors control, confirmation recognized data, etc.....

    €235 (Avg Bid)
    €235 Gns Bud
    38 bud

    **Description: We are seeking an experienced engineering consultant to conduct an in-depth gap analysis between our European-certified prefab walls and the 2018 (and beyond) ICC standards. This role involves not only comparing the two sets of standards but also converting the documentation to local US standards. The consultant will identify where our European specs exceed ICC expectations, where they fall short, and propose viable remedies for areas of non-compliance. **Core Responsibilities: - Gap Analysis and Standards Conversion: Lead a comprehensive gap analysis, including the conversion of measurements from metric to imperial and temperature from Celsius to Fahrenheit or Kelvin, ensuring accurate comparisons between European standards and ICC requirements. - Assessment of Compliance...

    €14 - €23 / hr
    Forseglet NDA
    €14 - €23 / hr
    5 bud

    ...engineer to implement an Opus encoder and decoder in C# for seamless integration with Asterisk. The project involves handling voice audio from Asterisk, decoding it, incorporating Text-to-Speech (TTS) functionality, and encoding the synthesized speech before sending it back to Asterisk. The ideal candidate should possess the following skills and experiences: - Proficiency in C# programming language. - Extensive experience with audio processing and Opus codec. - Familiarity with Asterisk, SIP, and IVR systems. - Knowledge of Text-to-Speech (TTS) integration. - Ability to deliver high-quality code within specified timelines. Main Tasks: - Implement Opus encoder and decoder in C#. - Integrate the solution with Asterisk for audio processing. - Inc...

    €1176 (Avg Bid)
    €1176 Gns Bud
    23 bud

    ...an experienced networking specialist to work as needed on various networks. Tasks will include: - Troubleshooting issues on existing networks - Designing and deploying new networks - Configuring switches and routers - Configuring wireless equipment - Securing networks - VPNs and remote access - Advising other team members on correct networking principles and processes - Troubleshoot issues with SIP and Video Conferencing Hardware includes PFSense, Sophos, Mikrotik, TP-Link Omada, Meraki, Cisco, Netgear and Dlink and others. First task will be troubleshooting a small business network in which network based software crashes when accessed from devices on the network, but doesn't crash when accessed from a virtual machine or terminal server on the same network. We have jun...

    €21 / hr (Avg Bid)
    €21 / hr Gns Bud
    30 bud

    In our quest for an outstanding and interactive VoIP web app, we're in need of an experienced developer, ideally one who is well-versed in Asterisk. The main features we're after, though not limited to, are call logging and reporting, an IVR system, and call routing and forwarding capabilities. Key Skills and Experience: - Extensive experience in VoIP development. - Proficiency in Asterisk. - Ability to develop an IVR system. - Experience in call routing, call forwarding, and call logging mechanisms. Given the nature of our project, it’s crucial to have a level of experience in these areas. Understanding the intricacies of these features is what will drive the success of our project. Though we have not specified it in the initial questions, potential co...

    €19 / hr (Avg Bid)
    €19 / hr Gns Bud
    15 bud

    I have install Asterisk (with freepbx), for gui to a Raspberry pi 4. I need a DEVELOPER TO WRITE a succesfull CODE and configurate the system to: Make a AUTO-VIDEOCALL from extention ''0'' to extention ''1'' or ''2'' or BOTH at the same time. The exctention ''1'' or ''2'' are log in to androip application, called ''PortSin'''or another app called ''Linphone''. Extention ''0'' is already loged into Raspberry with the programm ''Linphone''. When i press a button connected to GPIO 21 i want the audiocall to be started for 9 seconds. If the call is anwsered I also need to OPEN the DOOR by pushing ...

    €105 (Avg Bid)
    €105 Gns Bud
    10 bud

    I'm seeking a skilled professional to set up a new VoIP asterisk system with primary features of call routing and call recording. Must have the ability to have custom caller ID for outgoing calls. This project entails configuration for a small scale operation with less than 10 users/phone lines. Key Job Requirements: - Proficiency in setting up and configuring VoIP systems - Outstanding knowledge of asterisk - Experience with call routing and call recording This task requires an efficient and effective approach, understanding the needs of a smaller user-base. Someone with a strong track record of setting up VoIP systems and the knowledge to troubleshoot potential issues would be perfect.

    €201 (Avg Bid)
    €201 Gns Bud
    12 bud

    We used Asterisk as our phone system And we used Flutter's Sip-UA as a client The communication platform is WebRTC The problem we have is that when the internet suffers a few packet losses during a call, the client leaves the channel and then it is completely silent until the call is disconnected. We simulate the same scenario with a softphone, after the internet is disconnected and reconnected, the call continues and there is no problem. Are there any friends who can guide me?

    €461 (Avg Bid)
    €461 Gns Bud
    14 bud