Asterisk wavecom trabalhos

...am looking to incorporate AI features into my call center system, specifically Vicidial and Asterisk. As these platforms form the core of our operation, it is essential that any alterations enhance our outlay without disrupting the existing structure. Key Aspects of the Project: - AI Implementation: Even though I haven't specified the exact AI features to integrate, I'm interested in potential focus areas such as speech recognition and transcription, natural language processing, or sentiment analysis. Proposals that offer comprehensive strategies addressing these or other AI fields will be highly considered. - Dual Integration: The AI features must be incorporated into both Vicidial and Asterisk, aligning and harmonizing their performances. - Efficiency Goal: ...

I am seeking a skilled developer to construct a simple call forwarding system using Asterisk. Key functionality required includes forwarding calls to several numbers. This function should be dynamic, with the ability to add or edit the numbers to which calls are forwarded via a user-friendly web interface. Moreover, the web interface should give users the power to customize the call routing settings. Ideal candidates should have significant experience in VOIP technology, Asterisk systems, and developing intuitive web interfaces.

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

I'm looking for an experienced software engineer or IT specialist who can assist me by setting up real-time voice transcription on the VitalPBX platform. Key requirements and details of the project: - Integration of Google Cloud Speech-to-Text API: As I have already started this process, the successful freelancer will need to continue this work and ensure a successful integration. -...needs to be configured in such a way that it activates during multi-user conference calls. Your expertise in real-time voice transcription systems, particularly on VitalPBX, and thorough understanding of Google Cloud Speech-to-Text API will be crucial to the success of this project. I need it real quick and the idea is simple, user A calls user B and the conversation should be displayed in as...

We are seeking an experienced freelancer to help integrate Asterisk with Node.js for our communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP...

...create exporters for custom metrics when needed and represent the data in grafana dashboards. Grafana should be embeded in the existing web app either as public dashboards or iframes. Security concerns are a MUST. Things to monitor: system stats (node exporter): cpu, ram , disk, network... usual processes like sshd, cron, systemd-timesyncd, rsyslog... solution specific: mysql, nginx, kamailio, asterisk,nodejs, php-fpm, redis-server... Own developed services need some custom metrics. We can provide scripts that will return the data when executed. Data is meant to be presented to non-it people as well as it people. Meant to be visually atractive and easy to understand. Less easier graphs rather than many complicated ones The deployment in the product must be via configuration f...

I am looking for a freelancer who can assist with configuring my Issabel Asterisk system; especifically OUT AUTO DIALLER The specific configuration changes I need include: - Dial plan modifications - Voicemail setup - Call routing adjustments I have a specific deadline for completing the configuration, so it is important that the freelancer can work within that timeframe. I am already familiar with the system, so I do not require additional training or documentation on maintaining the new configuration. Ideal Skills and Experience: - Strong knowledge and experience with Issabel Asterisk system - Proficiency in dial plan modifications, voicemail setup, and call routing adjustments - Ability to work within a specific deadline If you have the necessary skills and experienc...

I am looking for an experienced freelancer to configure my Issabel Asterisk system. The project involves setting up specific features and implementing a detailed call flow. The ideal candidate should have expertise in call routing and be able to work with a call volume of less than 100 calls daily. Skills and experience required for this job: - Proficiency in Issabel Asterisk configuration - Strong knowledge of call routing - Ability to implement a detailed call flow - Experience working with a low call volume environment If you have the necessary skills and experience, please submit your proposal outlining your approach to configuring Issabel Asterisk and your previous relevant projects.

I'm currently in search of an experienced developer who can deliver a seamless integration of personalised ID setup on my existing VoIP services. I currently use 3CX, Asterisk, Vicidial, and Eyebeam for my business communications. Key Requirements: - Significant experience working with VoIP services including 3CX, Asterisk, Vicidial, and Eyebeam. - Proven expertise in VoIP system integration. - Ability to test for and ensure smooth communication across all platforms upon completion. Candidates with a strong background in VoIP software/platforms and successful project completions in similar setups will jump to the top of my list. Looking forward for a professional with a solid track record who can deliver an efficient solution.

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

We are seeking a skilled freelancer with expertise in Asterisk to assist us in implementing outbound call functionality to a mobile number using PJSIP. The current channel originate command is not working. and we are open to alternative methods that successfully initiate calls. Earlier, when we used older version of asterisk 18.1, we can get calls but newer version doesnt have that command and it gives below error pi*CLI> originate sip/ application Playback hello-world No such command 'originate sip/ application Playback hello-world' (type 'core show help originate sip/' for other possible commands) We are using asterisk 20.5 and system is rpi with ubuntu 23 installed. Thank you!

I'm seeking an expert with a deep understanding of Asterisk and Magnus Billing Portal to resolve a critical issue we're facing. Our system currently fails to establish connections for incoming calls right at the beginning of the call process. This persistent problem is resulting in missed communications and potential business losses. Ideal Candidate: - Proficient with Asterisk PBX systems - Experienced with Magnus Billing setups - Strong diagnostic and problem-solving skills Scope of Work: - Analyze and pinpoint the cause of the call connection failure - Troubleshoot and fix the routing issue to ensure call connectivity - Test and confirm that the issue is fully resolved Timeline & Communication: - Immediate availability preferred - Consistent updates require...

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.

Remove DND Button Add Login/Logout Button (Asterisk Queue) Add Pause/Unpause Button (Asterisk Queue) Add our logo Show Waiting Calls in Queue Show Agent Pause Status Show Agent Pause Count Show Agent Pause Time

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.

We need to connect our CRM with our phone system in place. We need integration between Zoho CRM and WAZO (Asterisk 15) Features: - Click to call from Zoho Interface: callback user extention + keep time duration - Pop up when incoming call from WAZO and open customer record or if customer is not registred, open record creation and keep call duration It worked with ZOHO Phone bridge before annd it worked proprely but ZOHO CRM stopped the service few years ago. Thank you

Responsibilities: • Develop and maintain Asterisk-based solutions for real-time transcription in an AI call center environment. • Integrate Whisper API for seamless speech-to-text and text-to-speech functionalities. • Implement real-time transcription capabilities for incoming customer calls. • Collaborate with the AI team to integrate intent recognition models for dynamic call routing. • Ensure smooth communication between Asterisk, Whisper API, and internal AI systems. • Optimize system performance to meet real-time transcription requirements. • Troubleshoot and resolve issues related to speech-to-text and text-to-speech integration. • Stay updated on advancements in Asterisk and speech recognition technologies. Requi...

...project of Asterisk call termination to WhatsApp. I have a cloud-based system for call termination and I require the specific call feature of call forwarding. I expect to handle more than 10 concurrent calls. The scenario is as follows: - The Asterisk PBX is hosted on AWS on Ubuntu 22.04 LTS - The asterisk PBX is able to make and receive calls already with the regular dial plan. Based on existing endpoints in conjunction with a Twilio elastic sip trunk. - As a second option we want Asterisk to dial a whatsapp endpoint. - We don't want any 3th party subscription options. - Any information on what Whatsapp Gateway service we can use appreciated. We prefer Twilio. But are not sure if Twilio is offering a Sip to Whatsapp Gateway service - You will build the...

In this project, you'll customise my Asterisk/FreePBX Laravel Panel to encompass a CRM module, following a corporate and professional design. Key features I'm looking for: - Call Recording - Call Forwarding - Interactive Voice Response (IVR) - Call recording The CRM module should: - Sync contacts and call history - Track sales leads and opportunities - Generate reports and analytics for user-friendly insights The ideal candidate for this project should have demonstrable experience in FreePBX, Laravel, and CRM integrations. An appreciation for corporate and professional design aesthetics will be a significant advantage. Your expertise will ensure fluid functionality and seamless integration, thereby optimizing our business operations. Project will be paid in mileston...

...provided with a text file containing 50 quiz questions with options. To clarify whether it is questions or answers, the questions are marked with question marks at the end of the sentence. The correct answers are indicated with asterisk in the file. The number of options can be different in each question. When the program runs, it will prompt users for specific parameters, including the number of questions to generate, whether questions should be randomized, and whether answer options should be randomized. The question must be displayed one after another. The asterisk must not included when displayed in the options in the output. When users answer questions incorrectly, the program will display the correct answers. At the end of the quiz, the total score will be presen...

I'm in need of an expert who can promptly install and accurately configure an Asterisk server for both inbound and outbound calls. Key Features Needed: - Call Forwarding - Call Recording - Call Routing - Dialer Calls Specific Call Recording Requirements: - Record all calls Your demonstrated experience in Asterisk server configuration, specifically with the above features, will significantly increase your chances of selection. A strong understanding of encryption is preferred but not crucial, as we require all calls to be recorded. You need to be detail-oriented in your approach, ensuring consistent uptime and high-quality call handling. Please bid based on the scope of this project, thank you.

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.

...want QTY >1) but they qty picker in the checkout seems to have an issue, the ajax doesnt always reload it to show new qty) Download files are not showing even though the database has them showing and I made a dummy download file for the dev site (this works before) but nothing is showing in user control panel. Payment method bullet points in cart need formatting to be darker as they are white. Asterisk needs moving a little on product page options (license options) mobile site - checkout spreads of the screen a bit. (Journal 3 checkout) Register form has been customised to replace the register form extension used on OC3 site, just some fields removed, this seems to work ok. I have had to change a few extensions as they did not offer OC4 versions, Bulli quick product edito...

Necesito configurar Asterisk en la versión enterprise de Pelo 17

I am looking for a developer who can open a link with caller and destination for every call via our Asterisk telephone system. Todo: We have A2billing and Freepbx (Asterisk). On each call open a URL like Example: Requirements: - Experience with Asterisk Ideal Skills: - Proficiency in Asterisk configuration and scripting If you have experience with integrating phone systems and web applications, please submit your proposal.

I have a rather old installation of ASTERISK (Asterisk 1.8.20.1) in a DELL r10 server. This is from 2013. I want to install a new version of ASTERISK (latest) and migrate all settings (one by one) in the new hardware. Job has to be done remotely.

I am looking for a freelancer who can create a Telegram bot that is linked to Asterisk and PBX. The bot should have both DTMF and Caller ID features. The primary purpose of the bot is to provide Sales Support. I am not sure about the specific requirements for the Asterisk and PBX integration, so I will need advice from the freelancer. Skills and Experience Required: - Experience in creating Telegram bots - Knowledge of Asterisk and PBX integration - Familiarity with DTMF and Caller ID features - Ability to provide advice and guidance on specific requirements for integration.

I'm in need of an Asterisk expert to configure a dialplan for my business. The core functionality required includes: - Attended Transfer & Whisper Transfer Dialplan: This specialized functionality should be available for agents / extensions . Skills and experience preferred: - Expert in Asterisk PBX configuration - Proven experience in setting up dialplans with the above-mentioned features. Your proposal should demonstrate your ability to carry out this project successfully. Note that the ability to meet timelines and provide post-delivery support will be a significant advantage.

Liderar una implementación de telefonía para 1000 internos. Se requiere experiencia en Asterisk y otros vendors.

We're in need of an Asterisk PBX expert capable of receiving text confirmation codes via SMS and converting them into multi-language voice IVR calls. Key Tasks: - Set up Asterisk system to receive SMS confirmation codes. - Convert the texts into voice IVR calls. - Ensure the system supports multiple languages/Libraries. - Forward calls to SIP trunk to a voice softswitch Ideal Skills and Experience: - Extensive experience with Asterisk PBX systems - Experience connecting Asterisk to SMS system to recieve the Text - Expertise in SMS/Text to voice conversion - Multi-language Library support capability - Background in telecommunication systems. In order to complete this project successfully, you should have a strong command of the Asterisk PBX s...

...freelancer who can help me set up a network solution infrastructure for my project. Here are the specific requirements and details: Asterisk Configuration 2 Web Server Configuration for WebRTC 3 Client-Side Implementation (Frontend) 4 WhatsApp Integration SSL Certificate installation 6 Testing and Debugging Network Solution: - I am looking for a wired network solution with Webrtc functionality. Existing Network Infrastructure: - I do not have any existing network infrastructure in place. Number of Devices: - Less than 10 devices will be connected to the network. Ideal Skills and Experience: - Experience in setting up wired networks and configuring Asterisk. - Proficiency in Webrtc technology. - Knowledge of network security and best practices. - Ability to troubleshoot...

Project Title: Asterisk Bluetooth - Adding New Features Platform: Linux Experience Level: Expert Project Description: - I am looking for an expert freelancer to enhance my existing Asterisk Bluetooth system by adding new features. - The purpose of this project is to expand the functionality of the system and make it more versatile. - The freelancer should have extensive experience working with Asterisk and Bluetooth technologies. - The main goal is to improve the overall performance and capabilities of the system. - The freelancer should be proficient in Linux, as the project will be developed on this platform. - Strong knowledge of Bluetooth protocols and audio streaming is required. - The freelancer should have a deep understanding of Asterisk configuration...

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I am looking for an Asterisk expert to provide technical support on a monthly basis. The ideal candidate will have experience in troubleshooting and maintenance of Asterisk systems at an intermediate level. You will be handling, maintaining and troubleshooting my billing, pbx and autodialer systems and will provide me support whenever needed. My offer is $100 per month. Specific tasks for the expert will include: - Troubleshooting any issues that arise with the Asterisk system - Performing regular maintenance tasks to ensure the system is running smoothly The complexity of the Asterisk system that needs support is intermediate, so the expert should have a strong understanding of Asterisk and its functionalities. The project requires more than 20 hours...

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 urgent need of a specialist with strong competencies in Asterisk and Javascript. The primary objective? Troubleshoot and fix a persisting playback issue with our Voice API's generated MP3 files. The successful candidate will apply their technical skills using any adequate tools to pinpoint the problem and implement a viable solution. Qualifying experts should possess: - Solid understanding of Asterisk and Javascript. - Proven experience with Voice API source code and file playback. - Exceptional troubleshooting skills, comfortable with using any debugging tools. - Capable of delivering practical solutions and testing for effective performance. Your role includes but is not limited to: - Delving into our API source code to identify the core problem preven...

Erstellt weden soll hier eine Android app mit AdMod " linephone " in verbindung mit asterisk oder elastix ,mit Funktion fax,sms location,

I am looking for a skilled developer to assist with the development of a Basic VoIP Hosted PBX system to work with SIP protocol with minimal featu...The project requires the following specific features: -Call forwarding -Call Blocking -IVR -Ring group -Extensions -Call Schedule & Call Queue - Call recording -Greetings - Auto attendant I also need guidance on specific integrations and customizations for the VoIP Hosted PBX. The preferred platform for the development is Asterisk. Ideal skills and experience for this project include: - Proficiency in VoIP technologies and protocols - Experience with Asterisk or similar platforms - Knowledge of call forwarding, call recording, and auto attendant features - Ability to provide guidance and recommendations for integration...

Complete Solution for GoIP Hardware (GSM Gateway + SIM Bank) Budget: More than $1000 Primary Purpose: Autom...software platform to manage a number of distributed network of GoIP GSM gateways and SIM Banks (SMB). We should be able to make mass calls, SMS/USSD bulk sending & receiving. Asterisk PBX must be used as a Core VoIP component. Features required: - bulk SIM/channel enable/disable/restart - IMEI display/modify - signal level display - human behaviour simulation (to increase SIM cards lifespan) - bulk SMS/USSD sending & receiving - automated SIM cards check, kick dead (blocked) SIM cards if detected - live tracking of ASR and ACD for each SIM card and channel - live total / answered / failed calls per SIM/channel Asterisk part must be built on top of PJSIP an...

I am looking for a FreeSWITCH specialist to assist with the installation and configuration of my VoIP system. Skills and Experience: - Extensive knowledge and experience with FreeSWITCH - Proven expertise in installation and configuration of VoIP systems - Strong troubleshooting skills to address any issues that may arise during the setup process Responsibilities: - Collaborate with me to understand my requirements and goals for the VoIP system - Install and configure FreeSWITCH according to the specifications - Test the system to ensure it is functioning properly - Provide ongoing maintenance and support after the installation and configuration process is complete If you are a FreeSWITCH specialist with experience in installation and configuration, please reach out to me. I am looking...

Hi there i need network administrator for sip to WP integration, i will send details too i have the clear of Asterisk. Let chat to disscus more."Sip provider

On a really old product server without any OS updates since about 8 years, the Asterisk server needs an additional functionality which is not available in that old Asterisk version. (Upgrading Debian or Asterisk isn't a solution as this server is running as it is and a rebuilt in any way is much too complex) On this server there are phone calls to given phone numbers initialized and the phone calls are recorded with Monitor. (MixMonitor unluckily is not included in my Asterisk version). The calls are managed in form of conferences. So only the mixed audio tracks (caller and callee) are recorded. But I need additionally the audio track of the callee seperately been recorded. And whatever I tried I was not able to do so. The problem seems to be that my does ...

I am looking for a freelancer to set up Kamallio as ...freelancer to set up Kamallio as a load balancer for inbound calls. This project is for an existing business and we require a round-robin algorithm for load distribution. We expect a call volume of more than 500 per day. In addition, we would like to have reporting features for analytics. A deployment and demonstration with all code edits of example application like from here - transcription app will be good. Ideal skills and experience for this job include: - Experience in setting up Kamallio as a load balancer - Knowledge of round-robin algorithms for load distribution - Familiarity with handling high call volumes - Ability to integrate reporting features for analytics

I am looking for a skilled developer who can help me with the development of a FreeSwitch + API VoIP system. The main functionality that I need for my system is call routing and forwarding. I do not have an existing VoIP system that needs to be integrated with FreeSwitch. For the API development, I am open to suggestions regarding the programming language to be used. Ideal skills and experience for this project include: - Strong knowledge and experience with FreeSwitch and VoIP systems - Proficiency in API development - Flexibility and ability to suggest the most suitable programming language for the API development - Familiarity with CRM integration would be a plus

Overview: This project aims to build an AI-powered voicemail system using either Twilio or Asterisk. The system will utilize advanced technologies like voice recognition, text-to-speech, and intelligent call processing to enhance the voicemail experience for Restaurant Baguette. Objectives: 1. To develop an interactive voicemail system with intelligent call routing and response capabilities. 2. Integration of voice recognition and text-to-speech technologies 3. To provide detailed documentation for system setup and troubleshooting. Call Flow: 1. Call answered on Twilio DID (or Asterisk vm). 2. Play recording: “You’ve reached the smart voicemail of Restaurant Baguette, how can I help?” 3. Forward call to “Vocode” () 4. Set up Vocode with Deep...

Hello, I want to work with WhatsApp, the solution should be a web page application based on sip+webrtc.1: User A make a call to whatsapp 2: Wahatsapp receive a text message 3: User A receive ringback 4: Whatsapp click on link and establish the video call are you able to do something like that Regards

...I'm looking for a knowledgeable individual based in the UK who can assist me in starting a VoIP service provider venture. I need guidance and expertise in setting up the infrastructure and initiating operations within the Asterisk PBX framework. **Key Deliverables:** - Consultation and guidance on establishing a VoIP service provider setup tailored to the UK market. - Configuration and setup guidance specifically within the Asterisk PBX framework. **Submission Requirements:** - Overview of previous experience in setting up VoIP service provider infrastructure or Asterisk PBX systems. - Proposed plan or strategy to assist in initiating the VoIP service provider setup. - Estimated timeline and pricing structure for consultation and support services. **Note t...

I am looking for a freelancer who can install Asterisk on my Ubuntu 20.04 LTS server for VoIP services. I have specific requirements for the configuration and customization of the Asterisk installation. Skills and Experience Needed: - Strong knowledge and experience in installing and configuring Asterisk on Ubuntu 20.04 LTS - Expertise in setting up VoIP services with Asterisk - Ability to understand and implement specific requirements for the Asterisk installation If you have experience with Asterisk installations and can meet these requirements, please submit your proposal.

I am looking for a skilled professional to help me set up a VoIP gateway with Asterisk. Integration with our own server is a must, so experience in this area is required. After the initial setup, I will require occasional troubleshooting support. Ideal skills and experience for this job include: - Strong knowledge of VoIP Gateways and Asterisk - Experience with server integration - Ability to provide occasional troubleshooting support

I am seeking a freelancer who can provide one-on-one lessons in VOIP, PBX, and Asterisk. As a beginner in this field, I am looking for guidance in setting up a PBX system, configuring Asterisk, and troubleshooting VOIP issues. The ideal freelancer should have advanced knowledge in these topics and be able to provide theory support and practical configuration instructions.