Asterisk Telephony

@kamran2012

Hello there, I will set up your Asterisk AMI click-to-call and click-to-hang-up flow, inbound PBX routing to agent mobiles, WebRTC via PJSIP with DTLS, and webhook-based real-time call data delivery. For your IVR chatbot integration, I will wire AGI scripts to handle STT/TTS handoff — keeping latency low by streaming audio chunks rather than waiting for full utterances before processing. Questions: 1) Which STT/TTS providers are you using — Google, AWS, or something else? 2) Is the Vicidial instance already configured, or does it need setup as well? Ready to start whenever you are. Kamran

@farooqtahir2013

As an astute Network, Cybersecurity, VoIP and System Engineer with an impressive decade-long experience, I am confident that I have what it takes to handle your Asterisk Telephony project adeptly. Having worked extensively with Asterisk and FreePBX systems, I am well-versed in all the aspects you've mentioned. From successfully implementing click-to-call functionalities for outbound calls to optimizing inbound PBX forwarding – I have tackled them all. Additionally, my proficiency in working with Asterisk Manager API (AMI) and STIR/SHAKEN framework will significantly contribute to your project's smooth execution; not forgetting my flair for dealing with crucial components like WebRTC, Sngrep, Webhooks and Web Sockets. Moreover, being experienced in Agile methods and committed to industry best practices, I guarantee reliable VoIP integrated systems that meet your unique business requirements. One of my recent projects even involved the implementation of an IVR system integrated with a Chatbot for a VoIP client - a task akin to what you need. This gave me remarkable insights into developing a seamless interface between different systems like yours. In essence, combining my extensive experience and skillset in Asterisk PBX, Linux, PHP and VoIP with your project's needs will ensure efficient communication solutions tailored just for you. Thank you for considering me and I look forward to working together!

@AZTURK1

Hi there, I will integrate click-to-call / click-to-hang-up flows using Asterisk AMI and FreePBX (Asterisk 13+) so outbound calls ring agent mobiles first, then bridge to the customer; I’ve deployed secure WebRTC (WSS + PJSIP+DTLS), sngrep monitoring and STIR/SHAKEN verification in similar PBX environments. - Implement AMI-driven outbound click-to-call and click-to-hang-up that dials agent mobile first, then customer; include AGI hooks for chatbot handoff to live agents (vicidial integration). - Configure inbound virtual-number routing to agent mobiles, IVR + speech-to-text / TTS integration and DB logging to MySQL; provide Webhook events via AMI/Dialplan. - Deploy WebRTC browser-phone (WSS) with PJSIP+DTLS, sngrep SIP monitoring, and STIR/SHAKEN attestation on outbound trunks. - QA & rollout: staged deployment, post-fix validation, rollback plan and minimal-downtime cutover; provide deployment notes and testing checklist. Skills: ✅ Asterisk PBX ✅ WebRTC (WSS), PJSIP ✅ AMI / AGI workflow and Webhooks ✅ Linux server deployment / FreePBX / SIP trunking ✅ VoIP security, STIR/SHAKEN, sngrep monitoring Certificates: ✅ Microsoft® Certified: MCSA | MCSE | MCT ✅ cPanel® & WHM Certified CWSA-2 I’m available to start immediately, deliver within 5 days for the initial integration. What SIP trunk/provider will you use for STIR/SHAKEN and outbound caller ID verification, and do you have existing TLS/DTLS certificates or should I provision them? Best regards,

@technologiesit

Drawing from my extensive experience in Full Stack Development with a specialization in Linux, MySQL, and PHP, I believe I am the ideal candidate for implementing the Asterisk Telephony project you have at hand. Having successfully worked on various IVR integration projects using Asterisk 13 and above as well as FreePBX, I have developed a keen understanding of various APIs (including AMI) and their applications like Click-to-call and Inbound PBX. Moreover, I have a strong background in setting up secure connections via WebRTC HTTP/WebSocketSecure (WSS), an element that your project also requires. My comprehensive grasp of Chabot integration including Speech-to-text and text-to-speech should prove valuable in building systems that effectively communicate with both agents and customers in real-time. With an emphasis on delivering scalable and secure solutions coupled with my tenacious dedication to client satisfaction as evidenced by over 100 successful projects, I am driven to leverage my skillset to make this project a resounding success for your business. Let's collaborate to not just meet your project goals but surpass them. Thank you for considering me.

@vishal1145

Your IVR-chatbot integration will fail under load if you're using AGI scripts synchronously - I've seen this crash systems at 50 concurrent calls when speech recognition latency spikes above 2 seconds. Before architecting the solution, I need clarity on two things: What's your current call volume during peak hours, and are you handling STIR/SHAKEN attestation at the carrier level or implementing it in Asterisk dialplan? This determines whether we need FastAGI with connection pooling or async AMI event handlers. Here's the architectural approach: - ASTERISK AMI: Build event-driven click-to-dial using originate commands with channel variables to track agent state, preventing double-dials when agents are already on calls. - WEBRTC + PJSIP: Configure DTLS-SRTP with proper certificate chains and implement ICE candidate handling to prevent one-way audio issues that plague 40% of browser-based softphones. - MYSQL OPTIMIZATION: Index CDR tables on calldate and disposition columns, implement partitioning for tables over 10M records to keep webhook queries under 100ms. - SPEECH RECOGNITION INTEGRATION: Use FastAGI instead of standard AGI to prevent blocking the dialplan thread, and implement timeout handlers for when Google/AWS speech APIs exceed 3-second response times. - VICIDIAL INTEGRATION: Set up proper lead recycling logic and agent pause code mapping to prevent leads from getting stuck in queue when chatbot transfers fail mid-conversation. I've built 4 production telephony systems handling 200K+ daily calls, including a healthcare IVR that achieved 99.7% uptime under HIPAA compliance. Let's schedule a 15-minute call to discuss your carrier's SIP trunk limitations and speech API rate limits before we start development.

@ashinnd84

Hello there, we are a team of capable developers and we can do this project in no time. Please, send me the project complete details to start the work. Thanks Ashish Kumar.

@NEHABHAT92

As an experienced professional with over 9+ years in web and mobile development, I am no stranger to the complexities of the IT world. My expertise in MySQL and PHP have proven invaluable in creating effective and sophisticated systems such as the one you require. In a previous project, I successfully integrated IVR—with speech-to-text and text-to-speech functionalities—and a chatbot system. Building upon this experience, I believe I can seamlessly incorporate your Asterisk Telephony needs as well. Not only do I possess the technical skills necessary for your project, but my approach goes beyond just delivering clean code. When you choose me, you gain access to a team that is not only readily available for support, but also committed to transforming your ideas into reality. My company's focus is on providing high-quality work while maintaining cost-effectiveness—a combination that sets us apart in the industry. In addition to these benefits, my proficiency in Android/iOS development presents exciting opportunities to further expand your project’s scope. Drawing from my toolbox of skills which run from Java to HTML/ HTML5/ CSS and more. All your current requirements will be met without any compromises on quality, delivery time or budget. Let's work together to make your Asterisk telephony system even better!

@Batainoronaldo9

With my solid years of experience as a Software Engineer, specializing in PHP and JavaScript development, I strongly believe I'm the perfect candidate for your Asterisk Telephony project. My expertise in working with Laravel, Vue.js, Node.js, MySQL, and more importantly, my proficiency in handling Linux-based systems are highly relevant to the tasks you need to be completed - be it Inbound PBX, dial plans, AMI usage, or WebRTC integration. Additionally, I have a good understanding of Click-to-call features from SIP signaling side. Moreover, my extensive experience in building scalable web applications, APIs and cross-platform solutions will prove beneficial in this project's multi-layered requirements. My strong suit in troubleshooting and problem-solving will ensure that any challenges faced in implementing Sngrep monitoring and integrating STIR/SHAKEN as validation layers are duly resolved. My affinity for innovation and forward-thinking aligns well with your project description. As a language enthusiast with conversational multilingual proficiency ( fluent in English & French), I can easily analyze and adapt to different API documentations if necessary. I am keen on joining forces with you to not only deliver an exceptional output but also provide great support post-completion. So let's get started on this exciting journey together!

@abdf2010

Hey, your project, Asterisk Telephony looks like a great fit for my skills. I've worked on similar PHP projects and can deliver solid results. Let me know if you'd like to chat about the approach.

@theavisharma

Hi, This is Avinash. I checked your requirement and understand you’re working on a complex Asterisk-based telephony system involving AMI, IVR, WebRTC, chatbot integration, and both inbound/outbound call flows. I can help you implement and stabilize features like click-to-call (agent-first then customer), inbound call routing via virtual numbers, and real-time event handling using AMI/webhooks. I’m also comfortable working with PJSIP, WebRTC (WSS + DTLS), and monitoring/debugging using tools like Sngrep. For your IVR + chatbot flow, I can assist in integrating speech-to-text / text-to-speech, managing call transfers to live agents (Vicidial/cloud), and ensuring smooth communication between all components. I focus on reliable call flow design, clean dialplan logic, and real-time system stability. Happy to review your current setup and help you move forward. JUST MSG ME.

@Royalitservices

Hi there, You’re absolutely in the RIGHT PLACE. I’ve delivered SIMILAR PROJECTS multiple times and know EXACTLY how to execute this efficiently and correctly from day one. To lock down the SCOPE, TIMELINE, AND PRICING, I’ll need to ask you a few key questions. Unfortunately, Freelancer’s 1500 CHARACTER LIMIT doesn’t allow me to break everything down properly here. Let’s jump on CHAT so I can show you my PROVEN PAST WORK, walk you through the REAL RESULTS I’ve delivered, and outline a CLEAR ACTION PLAN for your project. You’ll immediately see why my approach is DIFFERENT and EFFECTIVE. If you’re serious about getting this done RIGHT, I’m ready to move forward. Looking forward to CONNECTING and WINNING TOGETHER. Cheers, Mayank Sahu

@artur1992123

Hi Rajendra,! I can deliver your Asterisk AMI click-to-call system fast and clean. the agent-first outbound flow (call agent mobile → bridge customer) is the right approach for non-softphone setups. and handling inbound simultaneously requires proper channel management and AMI event parsing. I've built VoIP automation systems and worked extensively with Asterisk/FreePBX integrations in telephony projects. Is a 2-day implementation with proper error handling and call state tracking. Since I'm building my Freelancer profile and need quality reviews, I can deliver this for ₹12 500 ($150 USD), minimal budget, maximum quality. You get production-ready code, clean AMI integration. fast turnaround. Ready to start immediately. Check my portfolio for backend automation work. Artur

@mihajlodjokic

Let me get it done, I’ll integrate Asterisk AMI for click-to-call/hang-up, configure inbound routing via virtual number, implement WebRTC with WSS/PJSIP, and set up Sngrep monitoring. I estimate completion in 5-7 days, leveraging my experience with Asterisk and FreePBX integrations. Here is how I would approach it: 1. Detailed AMI configuration for outbound click-to-call functionality. 2. Virtual number setup and inbound call forwarding to agent mobile. 3. WebRTC WSS/PJSIP implementation with DTLS encryption. 4. Sngrep integration for call monitoring. 5. Webhook configuration for real-time call data transmission. I can provide a demo upon request. Let’s schedule a call to discuss specifics. Best Regards, Mihajlo

@himanshus642

Hi, This is a solid telecom + AI integration setup, and I can help you implement and optimize it end-to-end with clean, scalable logic. I’ve worked on: - Asterisk (AMI, AGI, dialplan) based systems - Click-to-call flows (agent-first, then customer dial) - WebRTC (WSS, PJSIP, DTLS) browser-based calling - AI IVR with STT/TTS + chatbot integrations ? Approach: - Click-to-Call (Outbound): AMI-based flow → call agent mobile first → on answer trigger customer dial (bridged cleanly) - Inbound Handling: Virtual DID → intelligent routing → forward to agent mobile with fallback logic - WebRTC: Secure browser phone using WSS + PJSIP (DTLS encryption) - Monitoring: Sngrep for SIP tracing + debugging call flows - Real-Time Data: Webhooks via AMI/AGI/dialplan to push call events (start, connect, end) - AI IVR Integration: Speech-to-text + text-to-speech pipeline Chatbot handling initial interaction → transfer to live agents (Vicidial compatible) - STIR/SHAKEN: Caller identity validation layer integration ? Delivery: - Working call flows (inbound + outbound) - Clean dialplan + scripts - Documentation + testing support I focus on stable, low-latency call handling with clean integrations across telephony and AI. Final budget can be decided after complete discussion. Let’s build this robustly ?

@ritasloves

As a seasoned developer with extensive hands-on experience in Asterisk, I can bring immense value to your project. My proficiency in PHP will help me not only understand but also adapt to your unique requirements which includes Click-to-call, Click-to-hang-up using Asterisk Manager api (AMI), understanding the inbound PBX and realise WebRTC – HTTP/WebSocketSecure for PJSIP with DTLS encryption. Moreover, I am well-versed in utilising tools like Sngrep and implementing Web Sockets – Web RTC based browser phone. Add trending STIR/SHAKEN authentication system – session border controllers and end-to-end encrypted media for identifying call sources, I'm confident of my abilities to handle all aspects of your project impeccably. With an existing project portfolio that includes IVR integrations, such as your ongoing project integrating Chatbot System using Asterisk, FreePBX and Speech-to-text and Text-to-speech features, I not only understand but have proven competence in the field. Couple this with my cloud operation (AWS, Google Cloud) expertise that ensures smooth-running of all operations and security, and your project couldn't be in better hands. Let's transform telecommunications together!

@sachendras2

I will design and develop a complete Asterisk-based VoIP call center solution including Click-to-Call (AMI), Click-to-Hangup, inbound PBX routing, and WebRTC browser calling with secure WSS and PJSIP (DTLS). I will also integrate real-time call tracking using webhooks, SIP monitoring with SNGREP, and build an AI-powered IVR system with Speech-to-Text and Text-to-Speech. Additionally, I will connect the system with Vicidial for live agent handling and ensure smooth outbound and inbound call flows using PHP, Linux, MySQL, and Asterisk PBX for a scalable and reliable communication system.

@techotd

As an accomplished team at TechOTD Solutions, we already have extensive experience in the field of Asterisk Telephony, particularly with AMI and managing PBX systems. In fact, currently on a similar project, we integrated IVR with Chatbot systems through Asterisk 13 and above versions. This allowed us to create a seamless user experience with efficient Click-to-call and Click-to-hang-up functionalities. Our expertise expands to working with WebRTC, ensuring encrypted communication (PJSIP with DTLS) while guaranteeing real-time call data syncing via Webhooks or WebSocketSecure (WSS). Familiarity with Sngrep enables us to actively monitor the SIP calls data, subsequently enhancing system transparency. Moreover, our skills extend beyond just achieving functional results; we prioritize creating future-ready solutions. In this context, matching incoming calls from virtual numbers to agents' mobiles is an area where we have proven dexterity. Adding Speech Recognition features is something we've already implemented for a previous client and making use of machine learning technologies like STIR/SHAKEN would be a natural transition for us. So, partnering with TechOTD Solutions equates to assuring long-term planning, scalability and ultimately, outcomes aligned perfectly with your expectations. Let's join forces to actualize your aspirations!