Asterisk pbx jobs

*** STRICT RULES - READ BEFORE BIDDING *** 1. DO NOT BID if you do not have direct, hands-on experience with ALL of these specific devices: - Cudy P2 5G Router - DrayTek Vigor 2915 Router - Yeastar S20 PBX I will NOT explain how these devices work or provide any tutorials. 2. ALL work MUST be done remotely via AnyDesk ONLY. No direct access, VPN credentials, or passwords will be handed over. 3. Do NOT message me asking "What is the problem?" or "What is your budget?". The entire problem, infrastructure, and scope of work are detailed precisely below. Your bid is your final price. Any messages asking these questions will result in an immediate BLOCK. --------------------------------------------------------- --- PROJECT DETAILS & SCOPE OF WORK --- We are look...

I need an experienced FreePBX professional who will teach me what I want to learn about fre...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 don't ...

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.

...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 prefer) along with the integration method for my existing CRM and ticketing tools. A brief outline of how you’ll handle testing and hand-over training will help me evaluate fit. Deliverables should include: • Fully configured call and WhatsApp workflows connected to my CRM and ticketing...

I need a working UAE local number that lets me place and receive calls from my Android phone exactly as if I were inside the country. Linkus is the reference point, yet I’m open to any SIP softphone or cloud-PBX solution that delivers equal or better results. Voice quality must be stable on both Wi-Fi and 4G, and the same local caller ID has to appear to the person I’m calling. SMS integration is essential—I must be able to send and receive texts on the very same UAE number through the chosen app or an associated API. Acceptable solutions may involve: • Provisioning a UAE DID, SIP trunk or virtual number. • Full configuration on an Android softphone (e.g., Linkus, Zoiper, Grandstream Wave, etc.). • Demonstrated test call and SMS exchange showin...

I have a brand-new Yeastar cloud instance already spun up and a Twilio account fully authorised for Australian numbers. My existing 3CX PBX is still live, but I now need everything cut across to Yeastar within the next 48 hours. Scope of work • Build the Twilio SIP trunk on Yeastar and confirm registration. • Create six extensions that mirror the current 3CX users (keep the same extension numbers). • Replicate the existing call routing rules—day/night logic, voicemail destinations, and any hunt or ring groups—so inbound calls land exactly where they do today. • Point each Twilio DID away from 3CX toward the new Yeastar trunk and verify inbound/outbound calling. • Run test calls with me on completion to confirm audio quality, caller-ID ...

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 technical details...

...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 want to wire Vapi’s Voice AI directly into my Yeastar P550 PBX so every outbound call to potential customers is handled automatically. The goal is true hands-free cold calling: the system should pick numbers from my lead list, place the call through the P550, open with an AI-generated script, hold a natural conversation, and either book the meeting or pass the call to a live agent when certain triggers are met. You’ll need to bridge Vapi’s API with the P550 via SIP or an equivalent method, build the call flow inside Yeastar (dial plans, queues, or IVR branches as required), and make sure the audio streams cleanly in both directions with minimal latency. I already have both Vapi and the PBX accounts active; what I don’t have is the middleware that le...

...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 starting over. Acceptance crite...

...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 testing, and final roll-out. Deliverables: • Fully configured Aster...

...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/out work straight away • Apply my specific FreePBX settings once I provide them • Supply root/GUI credentials plus a brief hand-off document outlining what you did and how to update the system in future Please factor in standard security best practices (firewall, fail2ban, S...

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

My 5G home connection is delivered through an Archer NX200 5G router and, like many 5G plans, it sits behind CGNAT. Because of this, I cannot reach my internal network from the outside world. The goal is clear: Remote Access. I need to log in reliably to my Computers, Smart Home Devices and Security Cameras, as well as a small PBX and a Draytek firewall. Dynamic DNS is already running with NOIP, yet CGNAT keeps every incoming request from ever reaching the LAN. What I’m looking for is a practical, reproducible way around that limitation: • A working solution that lets me initiate inbound connections (HTTPS, SSH, VoIP, etc.) from anywhere. • Detailed, step-by-step configuration on the Archer NX200 5G and any additional hardware, tunnel, or cloud service you ...

I need an expert who knows the Panasonic KX-NS100 inside out to take over ongoing maintenance and, more urgently, to hunt down stubborn routing problems. At the moment we are plagued by ghost calls, fake trunk activity and sporadic audio drop-outs that confuse users and clog the logs. These symptoms point to deeper call-routing faults, so I’m looking for someone who can dive straight into ARS tables, COS settings, SIP trunk parameters and firmware quirks to pin down the real cause, clean it up and stop the phantom traffic for good. You’ll be given remote access via VPN or TeamViewer during our business hours so you can trace packets, watch live call traffic and adjust the configuration in real time. Once the issues are eliminated I’d like concise documentation of every...

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...

...MVP) Metodología: Desarrollo Ágil / Entrega basada en hitos (Milestones) 1. Visión General del Proyecto y Arquitectura Objetivo Buscamos construir una aplicación descentralizada (dApp) en arquitectura Web2.5 que requiere un framework de comunicación de voz peer-to-peer (P2P) en tiempo real basado 100% en internet. El sistema debe omitir completamente el modelo de telefonía tradicional (PSTN/SIP/PBX), reemplazando los números telefónicos convencionales por direcciones de billetera cripto / Identificadores Descentralizados (DIDs). El acceso a los canales de audio debe estar estrictamente restringido por tokens / roles (token-gated) mediante el estado de contratos inteligentes o firmas en el backend Web2.5, garantizando der...

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.

Customer Portal & Admin Portal Requirements Project Overview Develop a modern, secure, and scalable multi-tenant Customer Portal and Admin Portal for managing Business Phone System (PBX/UCaaS) services. The platform should provide customers with a self-service experience while giving administrators complete control over provisioning, billing, support, and system management. Customer Portal The Customer Portal should allow customers to: Manage Business Phone System (PBX/UCaaS) services. View and manage extensions, users, departments, ring groups, queues, IVRs, voicemail, call routing, and other PBX features. Manage Direct Inward Dial (DID) numbers. Configure call forwarding, caller ID, business hours, time conditions, and other telephony features. Auto-provision...

...media company. We connect advertisers with retail media across multiple retailers through one point of access, and we give retailers the tools to sell and manage their own media offering. Tagline: "Retail media, made easy." Our audience is consumer brands, media agencies and retailers in the Nordics, with wider European ambitions. The symbol is already decided We have our logomark: the fine-line asterisk shown in the attached file. It is not up for redesign, and you do not need to propose a new one. Your job is to build the rest of the identity around it. You may adjust its scale, line weight and optical alignment so it sits correctly with the type, but the form itself stays as it is. Important: the name The full company name today is Retail Media Networks, but the weig...

...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 PBX deployment for a small office with seven new PoE IP phones that still require initial provisioning. We will handle the wiring part. You handle the software remotely. Scope of work • Install and configure the PBX on the platform. • Register all seven IP phones, apply the correct extensions, and confirm they can dial in, dial out, transfer, and park calls. • Enable the advanced features we know we’ll use right away: voicemail, call-routing rules (day/night, hunt groups, etc.), and conference calling. • Upload or record our custom audio prompts (welcome greeting, on-hold music). • Run end-to-end testing, then document the final settings so we can manage minor changes ourselves. Acceptance will be based on: – Every ha...

...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 showing me where to adjust lists, dispositions, and recordings. A quick walk-through over Zoom at the end will confirm everything is operational from soft-phones and, if possible, a physical SIP handset. Deliverable is a fully functional ViciDial dashboard that dials...

I am moving a small team (fewer than ten extensions) from our current cloud-based PBX to 3CX and want the changeover to be seamless. The help I need centres on two areas: • Data transfer – bringing across call records, phonebooks, voicemails and any other assets that 3CX can import. • Configuration and setup – installing the latest stable 3CX version on our chosen host, mapping SIP trunks, creating extensions, setting IVRs, ring groups and basic call-flow logic so that everything works exactly as it does today (or better). If you have ideas for user training, feel free to mention them, but it is not part of the core scope. Acceptance criteria • All existing numbers route correctly through 3CX. • Voicemail, call history and contacts are presen...

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all det...pay clients for their incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" includin...

...System Start‑Up,(Remote|USA, Our company is going to deliver automated wellbeing calls for seniors and vulnerable groups across the USA, We are scaling outbound capacity to 10,000+ calls/day and require specialists to implement the full technical and compliance‑ready system.260 cllents or more minimum start What We Need to Target. Core Skill Domains Dialer / outbound calling systems (VICIdial, Asterisk, Twilio, Plivo, Genesys, Five9, Talkdesk)Low sales staff for training remote. USA domilciled applicants only.3 speciallists at $2000 each for following 1 manage staff sales. dialer methods for assured pay costs once dialer documentation 2 Arrange subsciption services and documentation. 3 Adminisration of back office and ensuring flow on

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

...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. • ...

...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 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...