Brian burt asterisk işler

Google Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.

Merhaba Brian S., phalcon ve vue.js ile çalışmalar yapiyor musunuz?

Merhaba Brian S., phalcon ve vue.js ile çalışmalar yapiyor musunuz?

Asterisk santrali c# dili kullanarak ve kaynak kodlarıyla beraber bir yönetim programı oluşturmak istiyoruz. Daha önce asterisk santrali üzerinde çalışmalar yapmış, konuya hakim kişiler ö santrali c# dili ile yapılacak görseller projemize eklenmiştir. Buradan inceleyebilirsiniz. Bu tarz bir yönetim paneli talep ediyoruz. Bu projede bizlere yardımcı olacak ve proje ile ilgilenen kişileri teklifleriyle birlikte bekliyoruz!

Merhaba, 40.000 Kullanıcı kapasiteli, eş zamanlı 10.000 çağrıyı kaldırabilecek Asterisk kurulumu için destek verebilecek bu konuda IP Telephony, Call Center, Voip, Network konularında tecrübeli çalışma arkadaşları arıyoruz.

Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.

Merhaba Brian B., profilinizi fark ettim ve size projemi teklif etmek isterim. İstediğiniz ayrıntıyı sohbet üzerinden tartışabiliriz.

Merhaba Eric Brian, profilinizi fark ettim ve size projemi teklif etmek isterim. İstediğiniz ayrıntıyı sohbet üzerinden tartışabiliriz.

Yusuf Bey Merhaba, STH lisanslı bir telekom firmasında teknik departman müdürüyüm , şöyle bir sorunumuz var daha önceden çalışmakta olduğumuz aculab prosody s üzerine kurulu voip sisteminden asterisk platformuna taşınmaya çalışıyoruz test amaçlı kurduğumuz server üzerinde denemeler yapıyoruz, şu anda çağrılarımızda bir sorun yok istediğimiz şekilde çalışıyor olmasına rağmen dtmf detection sorunumuz var, ne yaptıysak asteriskin dtmfleri yakalamasını bir türlü başaramadık, bu konuda bize yardımcı olma şansınız var mıdır?Teşekkürler.

Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

Asterisk üzerinde çalışan crm yazılımını geliştirmek için proje desteği istiyoruz yapmış olduğumuz crm yazılımına data arama ve data kayıtlarının tutulduğu arayüz entegrasyonu yapılıcak.

...each agent individually: - Other agents cannot see each other's campaigns. 4. Ability for the main agent to assign one campaign to all of their agents. 5. Exporting statistics by agents: - Export a text (or any other) file with answered/unanswered numbers specifically for that agent. - Export statistics for a specific campaign or overall for the day. 6. Integration with Magnus Billing based on Asterisk 13: - Login credentials for the auto-dialing module correspond to Magnus Billing users. 7. It will be web-based. 8. Easy connection to another Magnus Billing server and transferring the module to another server. 9. Ability to manage AutoDialer users Create a new user, delete an old one, disable a user Please provide a brief overview of your relevant experience and how you...

...As we fire up our grills and make vegetable salads for our summer barbecues, we know that we can safely consume cured meats and veggies as long as we maintain moderation. The preservatives, both natural and synthetic, help keep our foods safe from harmful pathogens so we can enjoy our meals without worry. The food label will state that there are “no nitrates or nitrites added,” but an asterisk will often lead to a fine-print addendum with the clarification, “except those naturally occurring in celery juice powder,” sea salt or a vegetable juice. As a result some “natural” or “organic” roast beef and turkey breast, or other products cured with sea salt, evaporated cane juice, potato starch, or natural flavorings or seasoning...

I am in need of a simple, yet powerful tool that will enable me to capture and analyze the IVR menu options of my clients' toll free numbers. This tool will serve the ...Currently one of my team member is dialing the client's phone number, listening to the IVR options and manually capturing the details as follows *Level 1* Press 1 for English Press 2 for Tamil Press 3 for Hindi *Level 2* Press 1 if you are an existing client Press 2 if you are calling us for the first time *Level 3* Press 1 for sales Press 2 for Accounts Press 3 for service We already have a asterisk dialler in place and have the capability to dial phone numbers and also record the conversation but we don't have the ability to listen to the IVR options, dial each option to drill down to each ...

I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.

I'm in need of a SIP and Asterisk expert to help troubleshoot issues with a SIP interconnect between two companies, focusing purely on voice communication. Key responsibilities include: - Identifying issues within the existing setup - Repairing and configuring the SIP and Asterisk system Ideal skills and experience include: - Proven experience with SIP and Asterisk systems - Strong troubleshooting abilities - Understanding of VoIP technologies - Excellent communication skills

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

I'm in need of a knowledgeable and experienced Asterisk expert for the implementation in our company We need an Asterisk specialist to help us set up a carousel of different numbers and provider accounts. The essence of the task, when calling, automatically should be substituted random number from our list. Details and scheme will be discussed individually

I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.

I'm seeking a proficient Asterisk developer who can build a custom PBX system with Issabela or something comparable. The PBX system requirements are as follows: - Less than 10 extensions. - Features such as call recording, auto-attendant, and call routing. - Compatible with VoIP phone lines. Ideal candidate should possess a deep expertise in Asterisk and a good understanding of VoIP technology. Knowledge of Issabela or similar software is an added advantage. Let's connect if you can guarantee a seamless and efficient communication system. For more information, sent the requerimients - The calls received would be answered by several receptionists who would log in/unlog in with their landline to receive the call. If there are more calls than receptionists, a message w...

As a sysadmin developer, I'm in need of an asterisk specialist to build a Docker Compose script or a bash script for an interactive vocal server. This project is multifaceted, carrying out outbound calls and saving responses in a database. Key responsibilities are: * Creation of an Interactive voice response (IVR) system. * Outbound calling function connected with my API for automated scheduling of phone calls. * MySQL database integration to securely store the recorded responses. For this assignment, it would be ideal if you have proficiency in using Asterisk, Docker Compose, API integration along with comprehensive database management skills It would be a cherry on top if you have prior experience constructing interactive vocal servers. Let's connect to disc...

I'm looking for a professional who can install Asterisk PBX to facilitate a call routing system. Key Requirements: - Asterisk PBX will function primarily as a call router, modifying incoming caller ID's to the outgoing trunk DIDs. - The system should handle both incoming and outgoing calls efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice tran...

Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe

I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..

...highly-skilled Asterisk VoIP specialist for a unique project. I require expertise in Asterisk integration, VoIP configuration, and particularly in IVR implementation. KEY REQUIREMENTS: - Asterisk Integration: Full integration set up and testing needs to be completed. - VoIP Configuration: This includes configuring phone lines, extensions, and all essential VoIP functions. - IVR Implementation: The main purpose of this project is the implementation of automated IVR menus. This needs to be user-friendly and functional. The system should be capable of handling less than 50 concurrent calls. IDEAL CANDIDATE: The successful freelancer must have outstanding experience in all mentioned areas with a focus on IVR implementation. Any proven record in developing low-volum...

I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this project deman...

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

Seeking an experienced Asterisk developer to optimize the call quality of our existing VoIP system within our business operations. The ideal candidate must: - Possess 3+ years of experience in the domain - Be proficient with programming languages, VoIP, Asterisk interfaces (ARI, AMI, AGI), SIP configuration, API integration, and webhooks - Have robust problem-solving skills to provide innovative solutions Key responsibilities will include scrutinizing our present setup, identifying weaknesses, and implementing improvements to enhance call quality. A solid understanding of business requirements is vital to ensure the VoIP system is modified to suit our operational needs. Interested candidates, please email your resumes and cover letters.

I am searching for an exceedingly talented graphic designer to handle a print design project. Your expertise in innovative design styling is required to create engaging visuals. Key responsibilities and re...Your ability to create visually pleasing images, while adhering strictly to the design requirements will be crucial. Basic information:- You are invited to a invitation only Family Office event on Thursday 13th June 024 4pm at 25 Fitzwilliam Place, Dublin Investor Presentations (4pm to 6pm) 6pm - Drinks Presentations by London listing Music Company - ATC Group plc (Adam Driscoll and Brian Message), NASDAQ listed Isotope Enricher ASP Isotopes (Paul Mann), AI Fund Isomer Capital (XXXX) and Privately owned Police Intelligence Business Chorus (Boyd Mulvey) RSVP - XX...

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...

...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 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.

...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...

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.

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.

As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.

I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50

We are looking for an engineer proficient with Raspberry Pi, as we are in need of developing a VoIP PBX system on a Raspberry Pi 3 Model B+ . The end goal includes the integration of specific features into the system such as: - Call Recording - Voicemail to Email - Conference Bridging - Additional bespoke features Your expertise should include not only Raspberry Pi but also Asterisk and RASPBX/FreePBX. We are aiming for a robust, stable, and user-friendly system with custom features tailored primarily to business needs. The successful contractor will be required to develop the system on his own Raspberry Pi and submit an IMG file for loading onto other Raspberry Pis. The successful contractor will be working with our software developers so bids from individual contractors only w...

I have a video that I need two Reel/Tiktok style videos created out of. I will post one that I've made myself just to give you an idea of what I have in mind, but please do something better/more creative than I can, as I am not the professional. Please make sure that the video is cropped to only have the person speaking in the video edit. If it's cropped to include Brian and Erica, that is fine, but we cannot have our clients in the video. The time stamps that I would like the reels created for are: The first video- 45:15-45:53 The second video- 1:00:11-1:00:40. Feel free to get creative and add in pictures/graphics as you see fit. Also, please make sure that these videos are captioned. If you have any questions for me, please feel free to message me :) Thank yo...

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.

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: ...

Hi Mohamed A., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

I need a ghostwriter to focus on the global salt industry, covering all aspects from the production methods to its historical significance and its impact on the economy. The book should: - Provide insight into different geographical regions, emphasizing the respective methods of salt production. - Highlight the historical significance of salt, explaining its ...impacts. The ideal candidate will have a strong background in historical research and environmental studies, with exceptional writing skills to present complex information in an engaging and easily digestible format. Experience in ghostwriting non-fiction and storytelling is a must. Deep understanding of industrial operations and environmental impacts will be highly advantageous. something like Brian Freedman new book, ...

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