Asterisk pbx jobs

...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-switches or an Asterisk/FreePBX environment. • Confirm channel limits and any burst options so I can scale call concurrency when required. • Walk me through a quick test call—both inbound and outbound—to prove audio quality and stability. Once everything works as promised, the engagement is complete. If you already have direct carrier agreements in place a...

...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 privilege tables, HT...

...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. A test call to a public number completes with two-way audio. 2. The call appears in Twilio’s live call log. 3. No &ldq...

...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 the PBX for p...

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...

I already have a Grandstream UCM-series IP PBX on site and reachable for remote work. The hardware is up, the , but the system is still running on factory defaults. I need a Grandstream-savvy engineer to jump in and complete the core setup: • Create and test all user extensions (SIP and any analog FXS ports that may be required). • Build a clean inbound route so external calls land where they should. • Design and activate a simple operator/IVR menu that greets callers, routes them to the right extension or ring group, and falls back to voicemail if unanswered. Please handle dial plans, basic voice prompts, time conditions, and any minor tweaks that are typically part of making the above work smoothly. I’ll supply extension lists, Remote access is available...

My 3CX installation is failing to retain new voicemails—nothing is written to the local store or to the SharePoint location I have already integrated, and no notification emails reach the users. What I know so far • Storage paths under “Settings › Voicemail” point to the intended dire...permissions, SIP/SMTP log changes, backup edits, or SharePoint connector tweaks—so I can replicate them later. Acceptance criteria • A new test voicemail is stored in the designated location and visible in the 3CX console. • The same voicemail lands in the linked SharePoint library. • An email with the .wav attachment successfully reaches an Exchange mailbox. You’ll have remote access to the PBX and SharePoint tenant; just let ...

...voicemails have suddenly stopped being stored on the server and are no longer reaching users’ mailboxes. The PBX hasn’t been updated recently, so this isn’t tied to a new build or hot-fix. What I’ve already covered: I verified the voicemail settings, restarted the whole system, and combed through the logs, yet I still can’t see any obvious misconfiguration or error that explains why messages vanish. Here’s what I need from you: • Pinpoint the exact cause of the voicemail failure (storage path, permissions, SMTP issues, service crashes—whatever the root happens to be). • Apply or walk me through the fix during a remote session so that new voicemails are both retained on the PBX and forwarded by email again. • ...

I'm looking for an experienced sales agent who can find a qualified buyer for my production-ready VoIP infrastructure. The system includes a fully configured VICIdial + Asterisk 18 deployment with predictive dialing, Press-1 IVR, multi-agent support, SIP trunk integration, security hardening, daily backups, and a deployment image for fast restoration. Your job is not to configure or manage the system. Your primary responsibility is identifying serious buyers, introducing them to me, and helping close the deal. Payment will be commission-based: 10% of the sale price. Additionally, if you sell at a higher price than expected, you'll earn bonus incentives. Please include: - Your experience in B2B sales or telecom/VoIP sales. - Any relevant connections or previous successfu...

...codecs, and NAT/firewall troubleshooting - Basic VoIP quality checks and call-flow validation This is a small but time-sensitive setup task. I’ll share the exact requirements, current environment, and expected call flow privately with shortlisted candidates. Please apply only if you have configured Twilio SIP trunks with 3CX before, or can clearly explain your experience with similar SIP trunk/PBX setups. **Screening Questions:** 1. Have you configured Twilio Elastic SIP Trunking with 3CX before? 2. Can you handle both local DID and toll-free number routing? 3. Are you available to start soon after requirements are confirmed? 4. What access/details do you usually need to complete a SIP trunk setup?...

...and configured by someone based in the United States or Canada. The trunks must support unlimited concurrent channels and come with local U.S. DID numbers so my agents can 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...

...overcome objections, close the order, and record each interaction in the CRM. Occasional follow-up or outbound courtesy calls may be required when a quote needs that extra push. Clear spoken English, a quiet workspace, a reliable headset or VoIP line, and a stable internet connection are essential because call quality directly affects conversion. Familiarity with standard call-center tools or cloud PBX platforms will help you hit the ground running. Deliverables: • Real-time handling of all assigned inbound sales calls • Brief call notes entered into the CRM before logging off each shift • End-of-day summary report highlighting hot leads, closed sales, and any issues that need escalation If you can combine a friendly tone with proven sales skills while w...

I’m offering the complete outbound VoIP infrastructure that powers my own existing call center. Everything is already installed, tuned, and running in production, so you can migrate it to a fresh server and be live the same day—no development, no piecemeal setup, and absolutely no middle-man mark-ups. What you’ll receive: • VICIdial with Asterisk 18 fully configured • Predictive Dialer workflow ready to launch campaigns instantly • Press-1 IVR, auto-dial, multi-agent support, and SIP trunk integration baked in • Security hardening, daily backup routine, and a deployment image for one-step restore The handover includes root access credentials, written deployment instructions, and a short remote walk-through so you can verify everything ...

I’m compiling a prospecting list of 500 accountancy businesses located only in Bedfordshire, Hertfordshire, Cambridgeshire and Buckinghamshire that have no live website. Your job is to 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...

## 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...

I’m sourcing a reliable SIP route into India that lets me present an open CLI so my team can display our approved Indian numbers on every outbound call. This is strictly for business communication and we usually sit between 11 – 50 concurrent channels, so the route has to stay stable under that load without stretching latency or PDD. Call recording is essential. Whether ...clean, stereo-capable feed that captures every second of each call. If you can supply: • A-law / G.711 (G.729 as a bonus) • Consistent ASR and ACD that you’re willing to share up front • Quick test account or demo balance so we can check voice quality and CLI flexibility …then let’s talk. I’ll review your rates, payment terms, and any interop details you have ...

Project Overview We require an experienced Alcatel-Lucent Enterprise PBX engineer to review, troubleshoot, and configure our Alcatel OmniPCX Enterprise (OXE) system. The current access level available to us is limited and does not provide administrator privileges, preventing us from performing critical system configuration changes. The selected engineer will be responsible for restoring administrative access (where possible), reviewing the current configuration, and implementing the required PBX changes. Scope of Work: 1. Administrator Access Recovery Review current system access permissions. Assess available administrative and maintenance access levels. Recover, reset, or restore administrator-level access. Verify successful administrative login and access to configurati...

...Public IP, and CGNAT environments. Outbound calls are working only on Public IP connections and 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 voi...

... 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 cli...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 PB...

...robust, fully command-line-based provider manager for domestic US call termination. The solution must allow me to: • Send local ANI traffic, toll-free (TFN) calls, and unrestricted wireless calls. • 100% CLI outbound calls. • Maintain call quality and availability at the level of the most popular retail carriers. The acceptance process will be simple: I will install your configuration on a test PBX, make and receive all three types of calls mentioned, and monitor stability for 24 hours. If everything works correctly, with caller ID displayed on screen and no audio issues, we will need 200 to 500 ports to use, and you will act as our intermediary. Please suggest and secure your preferred carrier; we only ask that they operate within the United States and that yo...

We are a healthcare organization in Saudi Arabia looking for an experienced consultant to review and impr...experienced consultant to review and improve our call center operations. Current issues include: * Missed calls * Abandoned calls * Queue performance problems * Ineffective call routing * Reporting and KPI visibility challenges We are prioritizing candidates with: - Healthcare call center experience in Saudi Arabia and the GCC region - Experience supporting Arabic-speaking environments - Expertise in PBX, queue management, call routing, and CRM integrations We are specifically seeking consultants who have diagnosed and improved real healthcare call center operations. General AI developers are not a focus for this role. Please provide examples of similar projects you have ...

I need an Avaya engineer to finish setting up our IP500 V2 system remotely through TeamViewer or AnyDesk. The hardware is already racked and patched; what’s left is the software configuration inside Avaya IP Office Manager. Here’s what I expect when we are done: • External lines – all six existing analogue PSTN trunks must be programmed for both inbound and outbound calling, placed in a hunt group so that if one line is busy the next picks up automatically. • IVR / Auto Attendant – I will supply the written welcome script. You’ll record or convert it to a WAV file, upload it, and build a simple menu (Press 1 Sales, 2 Support, 3 Accounting). • User permissions (ARS) – three clear calling tiers must be enforced: internal-only, local-on...

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 URL that shows the exact resu...

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

I need my Zoho One environment to talk seamlessly with both WhatsApp and a cloud telephony platform. Within WhatsApp I must be able to send and receive messages through approved templates and free-form chats, and trigger fully automated responses that pull data from Zoho CRM, Desk and Campaigns. On the voice side, the solution must plug a cloud PBX into Zoho so every inbound or outbound call is logged automatically, recordings are stored against the contact, and IVR flows can pop lead or ticket details for agents. I also want real-time and historical call analytics surfaced inside Zoho Analytics. Because we serve opt-in customers only, an effective DND layer is essential; callers flagged as “Do Not Disturb” in compliance fields should be blocked from outbound voice o...

...lets my own custom CRM handle the following in real time: - Call logging (push every inbound / outbound leg with meta-data) - Click-to-call (simple HTTP/REST trigger that dials through FreePBX) - Call analytics (summary data we can poll for dashboards) The call features must be fully tested from the CRM side before delivery. Source code, endpoint documentation, and basic hardening of the PBX (fail2ban, strong passwords, TLS where possible) are expected. If you think a lightweight IVR is required for the trunk to work smoothly, let me know; otherwise no IVR is planned right now. I can give you immediate SSH access and will be available for quick test calls. Once everything is stable and documented, the job is done....

...multi-line dialer so prospects move seamlessly from list upload to a live transfer with my sales team. Proven experience with is non-negotiable. I would like to see the exact flow you have already deployed: how you sanitize and segment CSVs, the way you connect the dialer to the voice agent, any middleware you use for DNC scrubbing, and how you trigger warm transfers into a cloud or on-prem PBX. Detail the supporting tools—CRM, Zapier, webhooks, Twilio or similar—so I can visualise every hand-off and fail-safe. Deliverables • A documented architecture diagram of the outbound stack • Step-by-step SOP from lead upload to live transfer • Config files, API keys mapped (obfuscated), and best-practice call settings inside • One recorded walkt...

"not custom Dial Pattern ,, its be side Dial Pattern" We use MicroSIP and Linkus most of our PBX Use Yeastar and Granstream and may of PBX. Our Dial Pattern is: if you want to call Egypt, you must remove the country code (+20******) to 70*****, and similarly for short numbers – the country code must be removed, and a '7' added automatically based on the chosen dial pattern. Also, from Egypt to Saudi, dial pattern +966 changes to 6055******. We need templates for these dial patterns with options to add or edit numbers dynamically, without displaying them directly in the dial screen "back scene" . It should intelligently detect the correct country based on smart input. Additionally: - Desktop versions (Windows/macOS) - Mobile versions (Andr...

...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% ...

...FreePBX is online, I need an intercom workflow that works seamlessly with our IP phones and softphones. Create the required extensions and enable call-waiting, call-forwarding, on-demand and automatic call recording, plus live call barging/monitoring. Everything should be accessible through the standard FreePBX admin interface so future tweaks are straightforward. The final step is to expose the PBX safely to the outside world. We have a static public IP; set up the necessary firewall rules, NAT and any SIP or PJ-SIP considerations so remote devices register reliably without compromising security. Key deliverables: • Clean FreePBX 16/17 installation on Linux • Fully functional intercom with IP phone / softphone support • Extensions configured with call wait...

...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...

We have an active SIP Trunk service from STC Saudi Arabia and a ready 3CX PBX system. The SIP Trunk has already been activated by STC, and we have the required configuration details. We need an experienced VoIP / SIP / 3CX specialist to complete the remote setup, routing, and testing. Scope of work: 1. Configure STC Saudi Arabia SIP Trunk on 3CX PBX 2. Set up inbound call routing 3. Set up outbound call routing 4. Configure and test the available 5 channels 5. Test incoming calls 6. Test outgoing calls 7. Check SIP registration and call logs 8. Troubleshoot NAT, firewall, codec, or routing issues if required 9. Provide a short handover note with the final configuration Important: This is not a full call center setup project. The SIP Trunk service is already active. We o...

...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 need a complete VoIP call-center setup that plugs seamlessly...recording, warm transfers and supervisor whisper/barge-in. • Straightforward agent interface—soft-phone, web-RTC or Bitrix24 dialler—so onboarding takes minutes, not days. • Clear documentation, admin training and a cost-projection that keeps total spend predictable once the system goes live. Deliverables 1. Architecture diagram and tech stack recommendation 2. Configured Bitrix24 PBX or third-party PBX connected to Bitrix24 via REST/web-hooks 3. Fully tested call flows, dashboards and AI modules 4. Agent/supervisor training session (recorded) and hand-off documentation I will provide the Bitrix24 account, branding assets and SIP trunk credentials; you bring the know-how to ma...

I need my Zoom Phone account turned into a polished business phone system. The priority is to build a virtual receptionist that can immediately answer calls, play a clear custom greeting I will supply, offer recorded information, and then transfer callers to the right extension or voi...receptionist live and tested • Custom greeting uploaded, volume-balanced, and assigned to the correct entry point • Call-routing rules in place (answer, transfer, provide recorded info) • Forwarding logic documented and activated once we choose the final schedule • A short hand-off guide or video explaining how to maintain and update the setup If you’ve built similar Zoom Phone or cloud PBX setups, let me know—screenshots or a brief description of previous w...

I need an experienced VoIP engineer to take a self-hosted PBX environment to the next level. You should feel equally at home inside VitalPBX, FusionPBX, Issabel—or be ready to recommend another open-source platform if it serves the brief better—because I want the freedom to switch or combine solutions as requirements evolve. The immediate goals are clear: • Design and fine-tune advanced call-routing logic so every inbound and outbound call reaches the right place with minimal delay. • Build granular, real-time reporting that lets me track performance by queue, agent, and campaign without relying on costly third-party services. • Bridge the PBX with my existing tools (CRM, help-desk, analytics, or any REST/Open API-friendly system) so data flows...

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.

...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.) provided it integrates cleanly with ElevenLabs for TTS/STT and can scale to several hundred outbound calls per day. The agent must sound convincingly human, handle interruptions, and gracefully utilize our calendly to schedule another 15 minute google meeting live to walk through our pitch deck or end the call when needed. Deliv...

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 OTPs and automated tex...

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...

My VICIdial cluster is already live, leads flow into the hopper correctly, and outbound calls are reaching the 3CX PBX through the existing SIP bridge. The sole obstacle is that the system fires only one call at a 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 ...

...entirely online. The critical point is that the provider must NOT require any ID or KYC verification; I simply don’t have those documents. Once live, callers placed on hold should hear music automatically. This is strictly for my own personal use, not for business or resale, so capacity can stay modest as long as quality is clear and latency low. I’m comfortable configuring a soft-phone or lightweight PBX myself if you supply the trunk details, but I’m equally happy to let you handle the setup end-to-end. Deliverables I expect: • Working SIP credentials that register from standard soft-phones (e.g., Zoiper, Linphone) without ID/KYC checks • Ability to both make and receive calls to Irish landlines and mobiles • Hold music that plays whenev...

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

...procurement. Scope • CCTV & Access Control: exact camera locations, fields of view, cabling paths, power points, and terminations on the security rack. • Network Infrastructure: floor-by-floor rack elevations, switch counts, patch panel layouts, cable tray routes, Wi-Fi access-point mounting points, data and voice outlets. • Remaining services (to be completed once the first two are approved): IP-PBX, public-address speakers, meeting-room A/V, and any additional data/voice points. Deliverable Package 1. Fully annotated floor plans in DWG (AutoCAD) or RVT (Revit) with layer discipline separation. 2. Rack elevation diagrams and single-line schematics in PDF and native format. 3. Device schedule (camera, AP, switch, speaker, etc.) cross-referenced to...

...waits for DTMF 1, then plays the second MP3 and ends the call. • When all numbers have been processed the bot returns a Telegram summary listing each number that was successfully called. Scope specifics you must cover • Fix the current issue: “Bot not making calls.” • Restrict outbound traffic to USA numbers only; no international dialing should be possible. • Any voice gateway (Twilio, Asterisk, FreeSWITCH, etc.) is fine so long as you wire it cleanly to the Telegram Bot API and document credentials/config in an env file. • Concurrency should be configurable so I can tune call throughput without hitting carrier limits. Deliverables 1. Fully working Telegram bot (source code) that follows the flow above. 2. Simple README co...

...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 IP). • Vicidial’s login screen should ...