Asterisk dtmf hung punët

Pershendetje Ardit, keni eksperience me Suite crm apo Vtiger crm? kam nevoje pernje konfigurim ne lidhje me crm dhe asterisk

I'm in need of an Asterisk developer experienced with Asterisk 13. The primary task is to configure the system as a VoIP server for internal communication. Key requirements: - Understand Asterisk 13 well - Strong experience in setting up and configuring VoIP servers The ideal candidate should have: - Proven experience in customizing Asterisk for similar purposes - Good understanding of VoIP technologies and protocols - Ability to work independently and deliver high-quality work on time Please note that the project involves setting up the system, configuring call flows and ensuring it's ready for internal use.

I require an experienced and proficient developer who is familiar with both WebRTC and Asterisk. The objective is to expand the functionality of my current Asterisk setup by integrating it with WebRTC. The specific features to be integrated are: -Voice Calling -Video Conferencing This project seeks to enhance the communication capabilities of my Asterisk setup, thus the ideal candidate would have considerable experience managing such integrations in the past. Sound knowledge of VoIP, SIP, and related technologies will be greatly appreciated. An understanding of the Asterisk framework and its API will be critical.

I am in need of an expert who can effectively integrate WebRTC and Asterisk to achieve real-time audio and video communication. This functionality is intended primarily for internal team members. Therefore, it should ensure: - Seamless and effective call routing and handling - Successful integration with our existing telephony systems - Provision of standard-definition quality for both audio and video communication The ideal candidate for this project has substantial experience with WebRTC and Asterisk, and has successfully executed similar integration projects in the past. This professional should also be knowledgeable about telephony systems and their integration for real-time communication capabilities.

...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...

...directly staffed. There are staff near by that'll oversee this, but this is not their primary role. I anticipate either using existing cabinets (with some refreshing... vinyl graphic stickers/wraps?) or purchasing new cabinets. - One location will need all items that are showcased, locked up. The other location, some items (like shirts, sweaters..etc) can be left in the open to maybe try on? Maybe hung from a wall or case mounted hangar? -There will need to be some form of storage to house stock in either the cabinet or space. -The focus isn't so much the cabinet/storage, but the design of the space. As mentioned, this is a small space and are small parts of a function passenger terminal. But it needs to represent a small store. Some thoughts: - Maybe some vintage p...

Hi Viet Hung, I hope this message finds you well . Earlier this year you wrote me a google script to make a call from an api and import the json results on the google sheets. It was working great until recently . I keep getting the following error message: Error Exception: Address unavailable: at Search(Code:63:32) If I share the script with you , would you please be able to look at it and let me know how much it would be fix the issue ?

...hanging weights shafts, all while rotating. The front page of this spreadsheet will be dedicated to receiving calculation inputs. These will include: - Waterwheel dimensions: This involves measurements of the waterwheel, such as diameter, mass, etc. - Water flow rate: The rate at which water will be flowing onto the waterwheel. - Weight dimensions: Measurements of the weights that will be hung from the shaft. - Newton meters at the paddle: The unit of measurement for the Newton meters required at the waterwheel paddle. The ideal freelancer for this project should have: - Proficiency in Excel and designing complex formulas - A basic understanding of physics, particularly Newton meters and their application - Excellent communication skills to understand the require...

...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 ...

Hello Hung, I hope you are well. I have 31 items for edit with you. Would you like me to start a new job?

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 would like a painting which will depict my final memories of my dad before he passed away. I will be able to provide a photo of the location and also photos of him which you'll need to use to replicate my memory of him if at all possible. I'd like the painting to be watercolour on canvas so it can be hung or framed on a mantle/sideboard. I'd like a decent level of realism and accuracy whilst portraying bright water colours. I can provide multiple pictures of the location at my parents house and explain the scene and setting of the day. This will mean a lot to me and the picture will remain with me for the rest of my life. - Medium: Watercolor is the preferred medium for this particular project. - Artistic Style: A realistic interpretation is desired, the landscap...

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 currently in need of a dedicated Python developer capable of writing a specialized python script. Using Google Cloud Speech-to-Text, the script needs to recognize certain words from ongoing voice communication. Upon recognition of specified words, the code will need to send DTMF commands via a SIM7600 or SIP/VOIP system. In order to accomplish this project's requirements: - Proficiency in Python 3.5 is a must, as the script will need to be written fully in this version of Python. - Familiarity with serial communication is key, as this will be the method through which the script interfaces with the SIM7600 or SIP/VOIP system. - A thorough understanding of Google Cloud Speech-to-Text API is essential given its integral role in detecting specified words from speech in ...

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...

Black and white comic-book-style pencil sketch, very "sketch-y", very Todd McFarlane, lots of lines and sketchiness to it, not "outlined", if you know what I mean, though light in...surrounding the character). Ultimate image is of a female - not buxom, but still comic-book, powerful, in command, strong. Mainly of her (it's for my daughter, she's 5). Attaching images that capture the style - finished product would hopefully be in-between these two - capturing the raw energy of the seated-thrown Spawn, and not needing the full background of the other one. It will ultimately be hung, so needs some dark lines to be visible. I can find a better picture of the female main character, but the style capability is what I'm most interested in finding. P...

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...

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

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.

I am a chain supplier. I would like to have away to simulate the loads on a weight and chain double hung window system. I am most interested on the load the chain will encounter when a window is opened rapidly and hits the top and stops abruptly.

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...

Okay. Then please summarize in a few sentences what I will get from you if I give you the job.

Okay. Then please summarize in a few sentences what I will get from you if I give you the job.

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.

Hi Viet Hung, new project is getting $20 average would you write it for $30 since I talked to you first.

I'm looking for an experienced professional who can develop a custom call center with specific features such as the ability to handle both inbound and outbound calls, capture DTMF in live calls, and conduct auto-robot calls. Key Requirements: - Inbound and Outbound Call Handling: The call center needs to be able to efficiently manage both incoming and outgoing calls. This includes routing, call waiting, and call transfer functionalities. - DTMF Capture in Live Calls: A crucial feature I need in the call center is the ability to capture DTMF tones in live calls. This is essential for certain interactive voice response (IVR) systems and automated customer service functions. - Auto-Robot Call Functionality/IVR : The software should be able to execute automated calls...

Book cover design Size: 6 x 9” The design idea The image of a shaman depicted as shackled, hung on a stake, and left to die would certainly encapsulate the theme of the novel. It symbolizes the idea that for humans to fully embrace their agency and take control of their destiny, they must break free from the constraints of superstition and outdated beliefs represented by the shaman. Title: Pray, Let The Shaman Die Author: Osmund Agbo At completion, I would need in addition to png, word, jpeg files, the source PSD/AI file

I'm looking for a professional bio writer to help me create a personal bio to showcase my expertise and skills on my existing website. The bio should be engaging and friendly, aiming to introduce me to visitors. Key Deliverables: - Professional summary bio that showcases my expertise and skills - Engaging and friendly tone to... These are the instructions I was sent: Hi Chris, We'd like to include you in the "Team" section of our new website that is currently being built out. I've attached an example of what Megan's brief bio looks like and would request that you provide a brief write up for yourself and a headshot image as well. We will likely be bringing a photographer onsite to take some pictures so don't get to hung up on finding just the ...

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...

I'm looking 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 Integr...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 to bring this p...

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.

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

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