Asterisk i jobs

I’m rolling out a new Asterisk-based PBX that spans both an on-prem rack server and a small cloud instance. I need a specialist who can take the project from a clean slate to a fully functional, production-ready system. Here’s what I’m expecting: • Complete installation of the latest stable Asterisk on each node (one local, one cloud) with SIP trunks registered and ready. • Seamless call routing between sites, extension dialing, voicemail, IVR, and basic queue logic operating flawlessly across the hybrid topology. • Secure configuration—TLS/SRTP, strong authentication, sensible firewall rules, fail-over settings between the two servers. • Real-time monitoring and alerting set up (Nagios, Zabbix, Grafana, or a tool you&...

Dinstar UC2000-VE-4T + FreePBX/Asterisk + Retell AI Voice Setup Project Description We are looking for an experienced individual VoIP/Asterisk freelancer to configure a small business telephone system. Hardware available: Dinstar UC2000-VE-4T LTE/VoLTE Gateway 2 mobile SIM cards initially 4 available gateway channels Required configuration: Install and configure FreePBX/Asterisk Configure the Dinstar UC2000-VE-4T with 2 SIM cards Create two completely independent inbound/outbound call routes Configure two remote SIP softphone users working from PCs/mobile phones Each user must make and receive calls through their assigned SIM/number Incoming caller ID should be passed correctly to the remote user Configure correct outbound routing so each user always cal...

I have a set of handwritten notes compiled in a single PDF that I need turned into a clean, typed PDF. The original pages include clear headings and bullet-point lists, and I want that structure kept intact in the final document. Please make sure every heading is formatted consistently and that any bullet lists appear exactly as they do on the page. Where the notes flag an important point—usually marked with an asterisk or underline—use bold text so those items stand out immediately. Other than that, no special styling is required; a simple, readable font such as Calibri or Times New Roman is fine. Your deliverable is: • A print-ready PDF that mirrors the original structure while being fully selectable, searchable text. Accuracy is critical...

...The observed sequence is: agent 101 becomes READY → `outbound_autodial=Y` → VICIdial should select an eligible lead → an auto-call should be created/sent → `` should determine that a call needs to be placed → Asterisk should originate the call. Instead, the debug output from `` shows `READY=1 agents=0 dial_level=0 calls to place=0`, meaning the script is seeing the agent as READY but ultimately determines that there are zero agents/calls available for dialing and therefore does not originate a call. No corresponding outbound Asterisk channel is created. We have also inspected the VICIdial database and found that the AU_SOLAR campaign is active, the campaign has leads available in its list, and the campaign has previously shown a `vicidial_auto_call...

I need an experienced FreePBX professional who will teach me what I want to learn about freepbx operation and backup You can share your freepbx installation or use mine. You will look over my current installation if needed and help me fix the issues so that I can run it smoothly. So the focus is strictly on troubleshooting and ongoing support rather than fresh installation. I plan to use your experience instead of AI support. If you routinely work with Asterisk/FreePBX CLI, have a solid feel for SIP tracing tools such as sngrep or Wireshark, and are comfortable working/teaching on production systems where downtime needs to stay minimal, you are exactly who I’m after. I would like to use a help convenient with my work hours so that I do...

I'm looking for an experienced Asterisk VoIP engineer to help with PBX configuration, specifically for call routing. Key requirements include: - Expertise in Asterisk PBX - Proficiency in configuring call routing, especially geographic routing - Experience with VoIP systems and network configurations Ideal skills and experience: - Background in telecommunications or related field - Previous projects involving Asterisk and PBX systems - Strong problem-solving skills and attention to detail Please provide examples of similar work done.

I run a service center and I’m ready to streamline two key touch-points at once: inbound customer support calls and WhatsApp messaging. I am not working on CRM and ticketing system, so the solution must plug directly into both of those platforms—ideally through secure APIs or webhooks—so every interaction is logged automatically without any manual data entry. I only need straightforward rule-based flows: IVR menus for phone calls, and button- or keyword-driven replies on WhatsApp. Natural-language understanding is not required; speed, reliability, and clear routing rules are the priorities. Please propose the tech stack you’ll use (Twilio, 360dialog, Asterisk, Vonage, Dialogflow CX in “no-NLP” mode, or any other you pref...

I have an existing Python/Asterisk outbound autodialer running on a Debian 12 VPS. The application and dashboard are already built. I need an experienced Asterisk/ARI/AMI/Python developer to finish integrating the existing components, NOT rebuild the project from scratch. The system currently places outbound calls through a VMGate SIP/mobile gateway. Calling is working. We have also confirmed working two-way media/audio capture, AMI MixMonitor recording, DTMF transmission, and speech-to-text/transcription. The main remaining task is to connect the existing IVR state-machine () to the working live audio/transcription and DTMF system so the dialer can react to IVR prompts in real time rather than relying on fixed delays or post-call transcription. Important tec...

...last_update_time updates, ensure agents remain properly active/READY without falsely becoming LAGGED, and verify that autodialing works normally. The solution must be a permanent code/configuration fix, not a workaround such as manually updating MySQL timestamps, disabling LAGGED detection, or masking the symptom. The freelancer should be experienced with VICIdial, PHP, MySQL/MariaDB, JavaScript/AJAX, Asterisk and VICIdial background Perl processes and should document exactly what was changed and why....

I need a reliable SIP trunking solution running on Asterisk under Linux. At the moment I’m unsure whether the server already in place can be salvaged or if it will be cleaner to start from scratch, so the first step will be a quick audit of what’s there. From that point we can move straight into installing / re-installing Asterisk, pointing it at my chosen SIP trunk provider, and fine-tuning the dial-plan so inbound and outbound calls flow smoothly. Security hardening (fail2ban, TLS / SRTP where supported), codec optimisation for solid call quality, and basic call-flow features such as voicemail, IVR and call recording should all be factored in. I’ll need everything documented well enough that I can tweak extensions or add trunks later without...

I have most of the mobile application ready; what I still need is a solid Asterisk-based voice layer that simply works every time a user places or receives a call. My priority is configuration and integration: connecting SIP trunks, wiring up DID numbers, mapping inbound and outbound routes, and ensuring that every two-way call is recorded and pushed securely to my chosen cloud storage bucket. The core stack will run on a Linux server with Asterisk at its heart. You’ll handle the dial-plan logic, trunk registration, codec selection, security hardening, and the APIs that let the mobile app trigger voice calls and fetch recording links. I already know the basic architecture I want, so we can move quickly into hands-on setup, iterative testin...

I need a fresh VPS management server spun up and configured from scratch, then have FreePBX installed, fully secured and connected to my account. The VPS itself is not yet provisioned, so I will rely on you to recommend the most appropriate OS—CentOS, Debian or Ubuntu—explain why, and then handle the entire build. I do have a few custom requirements inside FreePBX (not a plain-vanilla install), so once the basic stack is running I will share the detailed dial-plan tweaks, trunk settings and any extra modules I need. Key deliverables: • Provision and harden the VPS with your recommended Linux distro • Install the latest stable FreePBX and Asterisk • Configure trunks, routes and SIP credentials for so calls in/o...

We are looking for a freelancer with hands-on experience in Asterisk, telephony, and SIP setup to support us on a short-term project.

I need a Telegram OTP bot integrated with Asterisk that includes a Caller ID feature. Key Requirements: - The bot should send and verify OTPs, as well as log OTP requests. - The Caller ID feature should display incoming caller numbers on just one device. Ideal Skills and Experience: - Experience with Asterisk and Telegram bot development - Familiarity with OTP generation and verification - Knowledge in implementing Caller ID functionalities

Require is a custom Asterisk-based autodialer engineered for high-volume, unregulated outbound traffic. The system must run on a hardened Linux distribution (e.g., Debian or CentOS) with Asterisk 18 or 20 compiled with specific modules: chansip for SIP trunking, appdial for call routing, and app_playback for IVR management. The core architecture includes a predictive dialing engine built in Python or C++ that interacts with Asterisk via AGI (Asterisk Gateway Interface). It must integrate the sofia or chansip channel driver to handle SIP trunks, supporting multiple concurrent legs for rapid call placement. For DTMF handling, the system will utilize the appdial with D option to detect dual-tone multi-frequency signals, routing them to specific IVR branches define...

M1: Outbound 407 diagnosis + VM fixes Full SIP trace of the failing outbound call, root-cause diagnosis of the 407, written trace-backed ticket for TeleCMI stating the carrier-side change required, and four Asterisk/PJSIP config bugs fixed on silent-sip-edge2. M2: Outbound live + AI pipeline handover Stable outbound SIP and RTP end to end, Silent Voice Kernel connected to the live call path, single pipeline with VOICE_MODE=FREE (Sarvam Bulbul v2) and VOICE_MODE=PREMIUM (ElevenLabs), test calls in both modes, handover notes.

...brands feel seamless together. Scope of work • One square Instagram post (1080 × 1080) • One Facebook feed post (1200 × 1500 or similar recommended size) • One email banner (1200 × 500) • Website hero banners for both desktop (3200× 1200) and mobile (1200 × 1600) Key messages – “Afterpay Day” – “15 % off* site-wide” – Clear call-to-action such as “Shop Now” – Include the small asterisk on “off*” for T&Cs Resources Brand site: Afterpay toolkit: Please keep copy minimal and let the bold type and colour speak. Final files should be supplied as layered PSD or AI plus web-ready PNG/JPG. Acceptance criteria 1. All dimensions correct ...

I need a specialist who can stand up an outbound ViciDial instance and connect it cleanly to Twilio so my customer-support team can start taking and placing calls right away. About 20 agents will log in daily, so the system must be sized and tuned for that load, including concurrent calls, proper carrier failover, and responsive web interfaces. Essential features • Call recording enabled on every leg of the call, saved with logical file names and dates. • Automated call distribution so incoming queues and outbound callbacks are routed evenly across agents, with simple skill-based routing rules. Your work will include installing or hardening ViciBox/Asterisk, configuring Twilio Elastic SIP Trunking, setting up the dial plan, test campaigns, agent log-ins, and s...

...total client payout, clicking on any line leads to detailed CDR of that result, this should be CSV downloadable, -Stats: Reports: Per date range, Per carrier total amounts. -Stats: Live calls, also with search options -Access lists: you know about this :) i will provide you a sample, and should be downloadable -Numbers which are not allocated to any client, they should play random IVRs from an IVR groups which we create and upload on the platform -Ability to download by the end of the month invoices for each carrier, with the form which i will send you. -Access list download: checking latest calls by date range and export file with range name, call from (country), call from (number calld minus 3-4 digits), last call date and time Client Interface: -CDR, amounts, reports:...

...circuits are busy now.” Nothing at all appears inside the Twilio call logs, so the traffic never reaches their side. I suspect the PJSIP trunk settings are to blame, though the outbound route or some subtle SIP-signaling mismatch could also be involved. No recent changes have been made to the trunk configuration, so whatever is wrong has likely been there from the start. What I need from you • Trace and isolate the fault across the outbound route, PJSIP trunk, and SIP signaling (sngrep or tcpdump, Asterisk CLI, etc.). • Apply the required corrections directly in FreePBX and confirm calls reach Twilio and complete successfully. • Provide a concise summary of the adjustments so I can reproduce them on other servers. Acceptance criteria 1...

I am ready to launch a new voice service and need a fully verified USA-based SIP trunk that lets me place and receive both domestic and international calls. Monthly traffic will sit in the 1,000 – 10,000-minute range and the trunk must support several concurrent calls without choking, so adequate channel capacity is essential. Here is what I need from you: • Introduce or provision a reputable provider that can activate an account in my company’s name with all necessary KYC completed. • Ensure the trunk is enabled for two-way voice, CLI presentation, 911 compliance, and international reach. • Supply full connection details (SIP URI, username/password or IP auth, codec list) plus concise step-by-step setup instructions for popular soft-switch...

...cleanly—Apache, MariaDB/MySQL, Asterisk and all VICIdial services report as running and phpMyAdmin shows the database online. Yet every agent now sees “Invalid Username/Password” on the campaign screen and the admin page just reloads when credentials are entered. Here is what I have already done: examined the Apache, MySQL, Asterisk and VICIdial logs for obvious errors, repaired and optimized the vicidial_users table, and confirmed the credentials being typed are correct. No browser-side tricks helped. We also do not have a usable database backup, so whatever fix you apply must leave the current data intact—no wipe-and-reinstall options are acceptable. I need you to: • Identify the exact point in the stack (MySQL privil...

...circuits are busy now.” Nothing at all appears inside the Twilio call logs, so the traffic never reaches their side. I suspect the PJSIP trunk settings are to blame, though the outbound route or some subtle SIP-signaling mismatch could also be involved. No recent changes have been made to the trunk configuration, so whatever is wrong has likely been there from the start. What I need from you • Trace and isolate the fault across the outbound route, PJSIP trunk, and SIP signaling (sngrep or tcpdump, Asterisk CLI, etc.). • Apply the required corrections directly in FreePBX and confirm calls reach Twilio and complete successfully. • Provide a concise summary of the adjustments so I can reproduce them on other servers. Acceptance crit...

Project Overview We are looking for an experienced FreePBX / Asterisk developer to install, configure, and develop a complete REST API for automating telephone bridge calls. We will provide: SSH access to a Linux server. SIP trunk credentials. Access to a remote server where call recordings will be stored. The final solution must be production-ready, stable, secure, and well documented. Phase 1 – Installation & Configuration The developer will: Install and configure FreePBX (preferred) or Asterisk if there is a clear technical advantage. Configure one or more SIP trunks. Our SIP provider allows multiple DDI (Caller IDs) on the same SIP trunk. The system must support multiple SIP trunks simultaneously. Configure automatic call recording. Secure and optimize th...

Design, configure, and maintain Asterisk-based telephony systems. Develop and manage SIP trunk configurations, IVRs, call routing, DID management, and dial plans. Integrate telephony systems with REST APIs, webhooks, and third-party platforms. Build and maintain API integrations for the Neurova platform. Configure inbound and outbound call flows, click-to-call, predictive dialing, and intelligent routing. Work with AI Voice Agents and ensure seamless human call transfer (human failover). Integrate WhatsApp, SMS, Email, CRM, and other communication channels where required. Troubleshoot voice quality, SIP signaling, RTP issues, and carrier connectivity. Collaborate with the AI development team to integrate Speech-to-Text (STT), Text-to-Speech (TTS), and LLM-based conversational workfl...

...place and receive calls without hitting channel caps or routing delays. Here’s what I’m expecting from you: • Active SIP trunk credentials ready for immediate integration with our existing PBX/soft-switch (we use standard SIP, so Broadsoft, FreePBX, Asterisk, or similar interoperability is fine). • A block of U.S. local numbers assigned to the trunk. • Confirmation that the service remains stable under sustained outbound volume typical of call-center traffic. • Basic caller-ID pass-through and a quick walk-through of any portal where I can self-manage future DIDs or channel additions. Acceptance criteria: trunks register successfully, inbound/outbound audio is clean, and at least five stress-test calls run simultaneously without q...

...research and populate a Google Sheets file with the fields below; those marked with an asterisk are absolutely mandatory: • Business Name* • Contact Number* • Companies House link* • Address • Contact Name • Email Address — this is the next most important item after the mandatory trio • Google Business Profile link* • Facebook Page link (when one exists) How you confirm the absence of a website is up to you; manual checks, directory scraping, built-in browser tools, or any combination that gets reliable results are all acceptable. Every entry must include a working Companies House URL and a callable phone number, and must genuinely lack an active site. Deliver the finished list in Google Sheets so I can review it...

## Brief: Multi-Tenant WebRTC-to-SIP Bridge for Asterisk Trunks Build a SIP gateway service that bridges WATI WhatsApp Calls API WebRTC media/signalling to standard SIP trunks usable by Asterisk. The system must support multiple tenants, where each tenant can have one or more WATI WebRTC endpoints mapped to one or more SIP trunks on Asterisk. The WATI integration uses SDP exchange over authenticated HTTPS APIs. WATI provides inbound call webhooks with `callId`, `sdp`, and `businessNumber`, and requires the bridge to generate a WebRTC SDP answer and POST it to `/calls/{callId}/accept`. Outbound calls require permission handling, SDP offer generation, and POST to `/calls/outbound-call/{waid}`. WATI requires Bearer-token authentication, Opus audio, complete ICE gatherin...

... No additional weights or styles are required. Characters Required The font must include all characters shown in the attached reference image. Uppercase: A-Z Lowercase: a-z Numbers: 0-9 Basic Punctuation: comma period colon semicolon exclamation question mark apostrophe quotation marks parentheses brackets braces hyphen en dash em dash slash backslash pipe at ampersand hash percent asterisk multiply divide equals tilde caret less than greater than Additional Characters: euro dollar pound yen section copyright registered trademark a umlaut o umlaut u umlaut eszett Final Deliverables After completion, provide: Editable font source file (Glyphs / FontLab / UFO or equivalent) Final font files: OTF TTF WOFF WOFF2 Complete working font with all required glyphs Font Met...

...require further assessment and optimization. Our objective is to build a scalable telephony architecture that allows customers to connect their existing phone numbers (Analog, SIP Trunk, Mobile, PRI/E1, PBX, etc.) to our AI voice platform. The consultant/freelancer should have strong experience in: •⁠ ⁠SIP, VoIP, PBX, and SBC technologies •⁠ ⁠Grandstream UCM, HT series gateways, Yeastar, 3CX, Asterisk, FreePBX •⁠ ⁠Session Border Controllers (SBC) •⁠ ⁠SIP Trunking and Carrier Integrations •⁠ ⁠NAT, CGNAT, TLS/SRTP, RTP troubleshooting •⁠ ⁠ElevenLabs, Retell AI, or similar AI voice platforms •⁠ ⁠Designing scalable multi-tenant voice architectures •⁠ ⁠High Availability (HA) telephony environments Current scale: •⁠ ⁠Approximat...

... No additional weights or styles are required. Characters Required The font must include all characters shown in the attached reference image. Uppercase: A-Z Lowercase: a-z Numbers: 0-9 Basic Punctuation: comma period colon semicolon exclamation question mark apostrophe quotation marks parentheses brackets braces hyphen en dash em dash slash backslash pipe at ampersand hash percent asterisk multiply divide equals tilde caret less than greater than Additional Characters: euro dollar pound yen section copyright registered trademark a umlaut o umlaut u umlaut eszett Final Deliverables After completion, provide: Editable font source file (Glyphs / FontLab / UFO or equivalent) Final font files: OTF TTF WOFF WOFF2 Complete working font with all required glyphs Font Met...

Looking for a skilled freelancer to design and develop a comprehensive panel for managing International Premium Rate Numbers (IPRN) services. The panel should include the following features: - Carrier management: Add carrier details, IPs/Ports, and personal notes. - Number management: Bulk upload numbers via CSV, allocate numbers to cl... country, rate, etc., with CSV export options. - Client management: Add client details (name, email, phone), allocate numbers to clients, and generate invoices. - Statistics: CDRs, live calls, reports on calls/minutes/payouts per carrier or client with CSV export options. - Secure platform with high capacity and quick installation script (.sh) including Asterisk/FS setup. Skills Required: PHP, Software Architecture, Asterisk PBX, ...

I need a developer who is equally comfortable scripting inside Asterisk and crafting a modern, clean web interface to turn my current False Answer Supervision (FAS) test scripts into a full-featured, client-facing tool. Here is what I want to end up with: • An automated tester that places configurable batches of calls through Asterisk, captures the raw CDRs and any SIP response codes, and stores them for analysis. • A web dashboard that presents the data in a crisp, modern layout and highlights two headline metrics I care about most: Call success rate and Error rate. Graphs, tables, and quick filters should make it easy to spot trouble routes at a glance. • From any report view I should be able to generate a public, read-only U...

Looking for VoIP/Asterisk expert to configure Yeastar device and integrate with AI voice we have: Yeastar TG200 LTE (firmware ), static LAN IP Vodafone CZ SIM with own phone number (LTE, excellent signal -67 dBm) Web admin access, API user configured account (BYO SIP trunk credential already created, but audio path doesn't work due to NAT/firewall) Network: standard LAN with NAT, server's public IP 185.21.220.66What we need: Outbound SMS via HTTP API (required) Identify correct Yeastar TG200 firmware 91.x API endpoint Working HTTP request (cURL or Node.js fetch) for sending SMS via Vodafone SIM Documentation: auth + payload + response Yeastar to SIP trunk (required - main task) Inbound (customer calls Vodafone number, then connects

...Engineering: Use specialized HTTP clients (like curl_cffi) to emulate proper TLS profiles and rotate residential proxy connections to bypass platform protections. 2. Telephony Backend & Carrier Integration: * Configure Asterisk to bundle 3 independent German SIP Trunks (Deutsche Telekom, Vodafone, o2 Telefónica). * The backend must use Asterisk Manager Interface (AMI) via TCP Socket to trigger simultaneous outbound calls to the seller's number when a match is found. * Multi-Trunk Routing: The system must fire parallel calls. The millisecond one carrier connects, Asterisk must send a clean CANCEL to the other channels. * Compliance: The architecture will organically broadcast the registered, real phone numbers assigned to my 3 trunks, ensuring 100% ...

...solution. Current Setup: - A GSM/VoIP gateway (Dinstar). - No physical server at the moment. - Planning to purchase a suitable cloud-based Linux server (e.g., VPS) based on your recommendation. - AnyDesk access is available for remote configuration. Scope of Work: 1. Recommend the best cloud server provider and required specifications suitable for our needs. 2. Install and configure VICIdial and Asterisk on the cloud server. 3. Integrate the Dinstar gateway with the cloud server. 4. Provide detailed guidance on connecting the Dinstar gateway with our router/switch, including cable instructions. 5. Configure inbound and outbound calling. 6. Set up agent logins and ensure web access for all users. 7. Test the complete system to ensure full operational capability. 8. Offer basic t...

...solution. Current Setup: - A GSM/VoIP gateway (Dinstar). - No physical server at the moment. - Planning to purchase a suitable cloud-based Linux server (e.g., VPS) based on your recommendation. - AnyDesk access is available for remote configuration. Scope of Work: 1. Recommend the best cloud server provider and required specifications suitable for our needs. 2. Install and configure VICIdial and Asterisk on the cloud server. 3. Integrate the Dinstar gateway with the cloud server. 4. Provide detailed guidance on connecting the Dinstar gateway with our router/switch, including cable instructions. 5. Configure inbound and outbound calling. 6. Set up agent logins and ensure web access for all users. 7. Test the complete system to ensure full operational capability. 8. Offer basic t...

...solution. Current Setup: - A GSM/VoIP gateway (Dinstar). - No physical server at the moment. - Planning to purchase a suitable cloud-based Linux server (e.g., VPS) based on your recommendation. - AnyDesk access is available for remote configuration. Scope of Work: 1. Recommend the best cloud server provider and required specifications suitable for our needs. 2. Install and configure VICIdial and Asterisk on the cloud server. 3. Integrate the Dinstar gateway with the cloud server. 4. Provide detailed guidance on connecting the Dinstar gateway with our router/switch, including cable instructions. 5. Configure inbound and outbound calling. 6. Set up agent logins and ensure web access for all users. 7. Test the complete system to ensure full operational capability. 8. Offer basic t...

I recently added a new SIP trunk to my setup—FreePBX 15 sitting behind a Kamailio SBC—but the trunk will not register and no outbound calls complete. I need someone who understands FreePBX 15, Asterisk SIP settings, and Kamailio routing to: • Diagnose why the trunk fails to register with the provider • Configure the correct authentication, codecs, and SIP headers so outbound calls connect cleanly • Provide a brief step-by-step summary of the changes you make so I can replicate them in the future You will have to work via Anydesk, you won't be able to directly connect to our server.

I want to deploy an AI-powered voice agent, built with ElevenLabs’ speech technology, that can automatically dial through our real-estate lead lists, hold a human-sounding conversation, qualify prospects and, when appropriate, lock in a phone consultation for one of our agents. Here’s the flow I’m envisioning: the bot connects via our dialer, introduces itself naturally, asks the key qualifying questions we already use, handles common objections, and—once the lead matches our criteria—books a follow-up call on the team’s calendar. All relevant call notes, recordings and contact details should feed straight into our CRM so no information is lost. You’re free to choose the back-end stack (Twilio, Asterisk, custom VoIP, etc.) provide...

Project Title: ​Expert needed to Integrate 4-Port GSM Gateway with Asterisk/FreePBX for Automated IVR & SMS API ​Project Description: ​"I am looking for a VoIP/Telephony expert to set up a complete automated IVR calling system for my fintech app. ​Hardware & Infrastructure: ​I have a 4-Port GSM Gateway (Dinstar/GoIP or equivalent). ​I have my own Cloud Server (VPS). ​Scope of Work: ​Asterisk/FreePBX Setup: Install and configure Asterisk or FreePBX on my cloud server. ​SIP Trunking: Securely connect my 4-Port GSM Gateway (located at my office) to the Cloud Server via SIP Trunk. ​Automated Calling API: Create an API that my mobile app can trigger to make automated outbound calls (Voice IVR). ​SMS Integration: Enable the ability to send O...

AI Call Center Developer – Twilio, Dialogflow, CRM Integration (Outbound Loans & Credit Cards) Responsibilities: Set up VoIP/telephony layer (Twilio or Asterisk). Implement Speech-to-Text (STT) for real-time transcription. Integrate NLP engine (Dialogflow or Rasa) for intent detection. Add Text-to-Speech (TTS) for natural voice responses. Connect system to Zoho/HubSpot CRM for lead logging and callbacks. Build branching conversation flows for multiple campaigns: Personal loans. Business loans. Credit card offers. Provide basic analytics dashboard for call outcomes. Requirements: Proven experience with Twilio, Asterisk, or similar VoIP systems. Strong skills in Python/Node.js for API integration. Familiarity with Dialogflow, Rasa, or NLP frameworks. Experience w...

...time, creating a severe bottleneck for my agents. I need the campaign and server-level dialer settings re-engineered so that the dialer can launch up to 20 simultaneous calls, keep the hopper continuously stocked, and eliminate any agent idle time. This assignment is a configuration task, not a hardware upgrade—the goal is to increase the dialing ratio and fine-tune any related rules that could limit call throughput. Key areas to address • Adjust the campaign’s dial level / adaptive or ratio settings to hit 20 concurrent dials • Verify hopper and list mix behaviour so leads feed without interruption • Confirm SIP and carrier limits are respected while preventing call queues or drop spikes • Document every parameter you change so I ca...

I need a dial plan that post messages to a website Have a IVR to read and update the website and database Log information and route based on situations Call back based on timeing Send calls based on timeing

I need a dial plan that post messages to a website Have a IVR to read and update the website and database Log information and route based on situations Call back based on timeing Send calls based on timeing

I already have a Telegram bot that should dial leads, play an IVR prompt, collect a “Press 1”, then play a closing message and hang up. Right now the Telegram side works, but the voice side does not—the bot simply isn’t making calls. Here is the exact flow I need brought to life (or repaired if you prefer to reuse my existing code): • I drop a .txt file of U.S. mobile numbers into the chat with the bot. • I send two MP3s: one to play on answer, the other after the caller presses 1. • The bot dials every number simultaneously (or as close as carrier limits allow), plays the first MP3, waits for DTMF 1, then plays the second MP3 and ends the call. • When all numbers have been processed the bot returns a Telegra...

My Vicidial instance on a Linode VPS has suddenly dropped off-line. The hostname no longer resolves and I cannot reach the machine even by its IP, so I’m convinced this is a network-level problem rather than a recent configuration change—nothing has been touched for weeks. I need someone who is comfortable diving straight into a production Linode, checking DNS records, firewall rules, networking interfaces, and Vicidial/asterisk services to get everything visible on the internet again. Root SSH will be provided the moment you’re ready to start; the goal is minimal downtime and a clear explanation of what went wrong so it doesn’t repeat. Deliverable: • The server’s web and SIP services must be reachable externally at (and by direct...

...mid-article Clean print stylesheet Section landing pages — three template variants for the Review's internal sections (Research, Essays, Signals). Each is a typographic listing page with a section title, a one-paragraph section description, and a reverse-chronological list of pieces. No hero image. See sample HTML for visual treatment. Issue page — template for displaying a curated issue (e.g., "Issue I"). Includes an editorial introduction from the editor, issue cover illustration, and a curated list of the pieces in the issue, grouped by section. Author / contributor page — template for individual contributors showing biographical information, photo, affiliation, and a list of their published pieces. Review About page — static page templ...

...Indian telecom regulations (TRAI/DoT). 2. Objectives • Automate bulk calling campaigns with minimal manual intervention. • Deliver pre recorded or text to speech greeting messages. • Provide a simple dashboard for campaign management, monitoring, and reporting. • Ensure scalability and reliability for large call volumes. 3. Scope of Work • System Setup: o Deploy open source dialer software (e.g., Asterisk, FreeSWITCH, VICIdial, GoAutoDial) or propose alternatives. o Integrate with SIP trunks/VoIP providers. • Features Required: o Upload and manage contact lists (CSV/Excel). o Schedule and batch calls to avoid congestion. o Play greeting messages (audio file or AI TTS). o Real time dashboard for call status (answered, busy, failed). o Reporting and an...

## 1. Project Scope The service provider will design, install, and configure a production-ready VoIP Call Center system using Asterisk on a Linux-based environment. The scope includes: * Server Environment: Installation and hardening of the latest stable Asterisk version on a cloud-based or local Linux server (Ubuntu/Debian). * Trunking & Connectivity: Integration of [X] number of SIP Trunks for external inbound and outbound calling. * Extension Management: Creation and configuration of [X] agent extensions supporting PJSIP/SIP protocols. * Call Flow Logic (Dialplan): * Development of a multi-level IVR (Auto-Attendant) menu. * Implementation of time-based routing (Business Hours vs. After Hours). * Queue Management (ACD): Setup of call queues with specific distributio...