Asterisk pbx jobs

...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...

Title: Complete Two-Mode AI Calling System Using TeleCMI, Asterisk, Sarvam and ElevenLabs Budget: Fixed price Duration: 1 to 3 days Type: Debugging, integration, testing and handover We need an experienced Asterisk and voice AI engineer to complete an existing outbound AI calling system for Silent Capital. Required architecture: TeleCMI PSTN to Asterisk 18.10 on a GCP Mumbai VM with static public IPv4 to Silent Voice Kernel to Deepgram Flux for speech recognition to Sarvam 30B for conversation to selected TTS provider to Asterisk to TeleCMI to customer The system must support two selectable modes using the same calling pipeline: FREE mode: Sarvam Bulbul v2 TTS PREMIUM mode: ElevenLabs using Mr. Kumar's cloned voice Switching must require only a conf...

...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...

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.