Freeswitch dialplantrabajos

Necesito Crear una interface Web C# mvc para recibir y Generar Llamados Utilizar WebRTC

buenas, e instalado astpp y le he echo todas las configuraciones, ahora tengo un problema y es cuando los clientes me mandan peticiones me cuelga el servidor, dejare la imagen anexa para que la vean

Somos una app para realizar llamadas internacionales y envío de mensajería de texto. Estamos en la búsqueda de un desarrollador o asesor que domine Ionic 4 para realizar integración para función de llamadas por internet con protocolo SIP y Freeswitch. La posición es para trabajar remotamente a tiempo parcial o brindar asesoría. Buscamos un profesional de alto nivel cuyo trabajo sea estructurado, en equipo y organizado. El trabajo será remoto, desde su casa, utilizando la herramienta de monitoreo de Hubstaff.

...una integración webphone de asterisk. Con la cual se pueda realizar llamadas de un área a otra, de una persona a otra, de un anexo al exterior. Entre las funcionalidades que debería contar Asterisk es: * Grabación de llamadas * Límitador de llamadas * Ocultamiento de número celular destino * Habilitación de soporte web para el servicio Asterisk * Habilitación de soporte webrtc * Programación de dialplan para click2call * Configuración de Troncal SIP con operador * Programación de soporte para anexos dinámicos y temporales * Creación de servicio web para generación de anexos temporales y dinámicos * Creación de script de consumo de servicio web para devolución de...

Requiero experto en Asterisk que maneje perfectamente los contextos, dialplan, CMD, scripts. Que conozca a detalle los equipos Linksys y GrandStream y su configuración avanzada como polaridad, tonos desconexion/colgado, frecuencia, voltajes en lineas ... Nivel avanzado en solución de problemas.

Requiero una persona con experiencia en el sistema de FreeSwitch "FusionPBX", para instalar en un server dedicado y configurar una seria de parámetros. Eventualmente cada mes se procederá a chequear que el server y los servicios están correctos.  Es indispensable conocer el FusionPBX ya que se necesita sobre todo activar la opción Multitenant, asi como colas de llamadas, IVR, BFL, y capturar llamadas.<br />Otro equipo llevará la getión diaria de los clientes, cuentas, y funcionamiento en general. <br />La finalidad de este proyecto es para la instalación y primeras configuraciones, asi como soporte eventual.

Instalación de Freeswitch-Fusionpbx en pcb Banana Pi, con Debian 8 ya instalado en tarjeta sdcard, sería a través de SSH ó teamviewer

, conocimiento u orientación profesional hacia el área de software libre y de código abierto (FOSS), administración de servidores Debian Linux, VoIP (Freeswitch), Moodle, desarrollo móvil, APIs, Think Client, herramientas de groupware, entre un profesional de tecnologías de información consciente de la importancia de la programación en Bash y herramientas orientadas a consola. En pocas palabras: amor por la "ventanita negra" o "línea de comandos" de UNIX o GNU/ de administración de servidores GNU/Linux (Debian), SSH, Shell y programación en Bash, aplicaciones en modo consola (vim, emacs, tmux, irssi, htop, etc.), configuración de redes y Thin Client, virtualización.4. ...

Necesito poner en el dialplan un ringback (se que en "Asterisk Trunk Dial Options" pongo la opcion "r"). Pero necesito poner que se reproduzca por maximo x segundos, sin detener la ejecución de la llamada (he intentado poner "wait()", pero detiene la salida de la llamada). Lo que busco es que al marcar se reproduzca el ring back falso y simultaneamente la llamada se tramite y a los X segungos se quite el ringback falso y deje el audio original que trae la llamada

Necesito conectar una planta Asterisk a una Planta Avaya por medio de una troncal H323. En este momento tenemos problemas de señalizacion y se pueden generar llamadas via Asterisk -> Avaya pero no vie Avaya -> Asterisk. Si tiene conocimientos en el manejo de otra PBX como Freeswitch o Yate, se puede usar esta tecnologisa y subirlo como Gateway.

Tenemos una plataforma desarrollada en freeswitch que ofrece servicios de audioconferncia. Buscamos personas que nos apoyen en el mantenimiento de dicha plataforma.

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Buenas tardes, Se necesita programador especializado en centralitas Asterisk utilizando Elastix, para configurar Elastix con extensiones Dahdi, crear IVR y otros servicios personalizados de Dialplan. Estamos buscando programador que trabaje en España, preferiblemente en Madrid para reuniones ocasionales Gracias.

Buenas tardes, Se necesita programador especializado en centralitas Asterisk utilizando Elastix, para configurar Elastix con extensiones Dahdi, crear IVR y otros servicios personalizados de Dialplan.

Se pretende migrar el actual sistema de telefonía basado en Cisco Callmanager a un sistema Asterisk. Son 200 extensiones de teléfono repartidas en 4 sedes. Se requiere implementar el dialplan y ayudar a la puesta en producción.

Creacion de CRM usando Asterisk, Freeswitch, php y mysql.

Se pretende migrar el actual sistema de telefonía basado en Cisco Callmanager a un sistema Asterisk. Son 200 extensiones de teléfono repartidas en 4 sedes. Se requiere implementar el dialplan y dar soporte en el despliegue en producción.

Necesito poner en el dialplan un ringback (se que en "Asterisk Trunk Dial Options" pongo la opcion "r"). Pero necesito poner que se reproduzca por maximo x segundos, sin detener la ejecución de la llamada (he intentado poner "wait()", pero detiene la salida de la llamada). Lo que busco es que al marcar se reproduzca el ring back falso y simultaneamente la llamada se tramite y a los X segungos se quite el ringback falso y deje el audio original que trae la llamada

...en servicios de comunicación de voz en mercados emergentes con un rápido crecimiento, ofreciendo a carriers calidad y alto rendimiento de la red a través de interconexiones directas. Estamos buscando a un Programador con experiencia que pueda crear aplicaciones para VOIP, conference calls, chat y streaming audio utilizando Asterisk y otras tecnologías de código abierto (por ejemplo OpenSIP, Freeswitch, SER, etc.) Responsabilidades: La persona elegida trabajará en un entorno dinámico y estimulante, desarrollando e implementando soluciones únicas y personalizadas para satisfacer las siempre cambiantes necesidades de la industria de las telecomunicaciones y a la vez aprovechar las oportunidades de mercado que se presentan. L...

Complete FusionPBX Setup and Configuration - dialplan outbound - outbound call - 2 did inbound - different group ring - announcement

I am seeking a highly skilled freelancer with intensive experience in FusionPBX and Freeswitch. The main objective of this project is the creation of a precise dial plan for FusionPBX. The developed dial plan should effectively handle active calls - holding them temporarily during paging calls. Key tasks include: * Dial plan must allow higher priority calls to override held calls * Acceptance of priority levels for different callers as input parameters The ideal candidate should possess these skills and experience: * Proven experience in FusionPBX and Freeswitch * Knowledge of implementing dial plans * Understanding of call flows and priority handling. Your task will revolve around ensuring a smooth call flow process during the paging, which detailed understanding of ...

I am in need of an expert OpenSIPS VOIP Developer who can help create a robust system with high quality call routing. The main feature's to be implemented are NAT Handling in the Cloud, Microsoft Direct Routing, and Mid Registrar. The system should be designed to interact efficiently with PBX systems (FreeSWITCH) and OpenSIPS. Required Skills: - Extensive experience with OpenSIPS VOIP system development - Strong understanding of direct call routing - Proficiency in integrating VOIP systems with PBX and OpenSIPS - Familiarity with general telecoms concepts and technologies. The ideal freelancer should have worked on similar projects and be ready to showcase relevant examples in their proposal. If this project goes well, there could be more ongoing work. I look forward to disc...

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

I require an expert in Asterisk and FreeSWITCH development, along with experienced system engineering skills. Specifically, I need help with: - Asterisk and FreeSWITCH development - Integrating VoIP - Setting up Call routing and IVR (Interactive Voice Response) - CTI - Skill group base call routing to agents. Apart from these, the implementation of Asterisk and FreeSWITCH clustering as well as Call Center Reporting is required. The technology stack should be Linux, PHP, MySQL along with postgresql and API knowledge. The freelancer should have considerable experience in these to meet the project's specific requirements.

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

I'm looking for an OpenSIPS expert with comprehensive knowledge in call routing. currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have iss... currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have issued when A call B, and B ring 180, after that B refer call to C, call not reach to C - I'm follow opensips Docs we have to config uac_redirect but i try with no luck 2. We use media server is Freeswitch, how can i pass a variable to Freeswitch, i try add header follow format: X-Variable, sip_h_X-Variable but in luascript of FS...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related applications

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

I have a particular project requiring expertise in Freeswitch, preferably with exposure to a Debian Linux environment. The key objective is to use Freeswitch for answering machine detection. The main requirement is to capture and analyze voicemails. We want to build a custom module or ready solution which can detect human or machine on the outbound dialer. The solution should work on any freeswitch version and without any licensing restriction. Please note, we have already used AVMD module and it just detects beep. We want to detect the machine or human from the initial screening of voice only. Skills and Experience Needed: - Solid experience with Freeswitch answering machine detection - Familiarity with working on Linux systems - Ability to effectively capt...

...Compatibility: Implement gRPC to ensure compatibility for seamless communication between components. 3)Real-time Transcoding: Enable real-time transcoding capabilities to convert audio data appropriately for interaction with OpenAI Whisper. 4)Asterisk Integration: Integrate the solution with Asterisk to allow seamless initiation and handling of audio calls from Asterisk dialplans. Example Asterisk Dialplan: [audiosocket] exten = s,1,Answer() same = n,AGI() same = n,MixMonitor(/var/www/html/calltest/${uuid}.wav) same = n,Ringing(3) same = n,Wait(1) ;same = n,Playback(silence/1) ;same = n,Playback(custom/audioStream) same = n,Dial(AudioSocket/${uuid}) same = n,Verbose(0,Result was ${SPEECH_TEXT(0)}) same = n,Hangup() exten = h,1,Hangup() Deliverables: 1)Fully functional

Task details - For ISUP/SS7 (E1 card implementation) along with IVR knowledge in any open source SIP servers like Freeswitch/Yate/Mobicents etc. or working experience in CRBT server. Interested candidates please apply

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

...freelancer who can assist us with configuring STIR/SHAKEN on our existing Freeswitch / ASTPP setup. Here are some details about the project: - The freelancer does not need to have the necessary certificates and keys for STIR/SHAKEN implementation as we will provide the necessary information. - Our Freeswitch / ASTPP configuration is already fully set up, but it needs modifications to support STIR/SHAKEN. - We are open to suggestions and do not have any specific requirements or preferences for the STIR/SHAKEN implementation. Ideal Skills and Experience: - Strong knowledge and experience with Freeswitch configuration. - Familiarity with STIR/SHAKEN implementation and protocols. - Ability to make modifications to the existing Freeswitch setup to support STIR/...

If you do not have intimate knowledge of fusionPBX/Freeswitch, do not waste your time. I need the following: Integrated GUI/modal for fusionpbx (php). 1. select campaign or create new () 2. define extension number and survey name ( and ) 3 allow user to config the survey with adding N questions. Each question should allow user to define a label, select audio from the uploaded audio files. Also each question should allow the user to specify allowed responses (dtmf button presses) 4. Have a section for output statistics, auto email option, export to pdf (pretty report and I will provide the mockup) and/or csv data. Note: I will provide mockups to this 5. This should also be integrated with the permissions so I can set permission levels for me and my customers

...custom API that interfaces with our FusionPBX and FreeSWITCH telephony system. The primary goal of this API is to manage telephony features programmatically, including customer management, DID handling, call routing, and call feature configurations. This project requires deep knowledge of FusionPBX, FreeSWITCH, and database management using MariaDB. Key Project Tasks: Customer Management: Create and update customer records, including status management (Active/Inactive). DID Management: Insert, update, delete, and route DIDs within the system. Call Feature Configuration: Enable/disable call recording for specific numbers, and set/modify caller IDs. CDR Synchronization: Develop a mechanism to synchronize Call Detail Records (CDR) from FreeSWITCH to a SQL-based applic...

...need of a professional to help with configuring FreeSWITCH and FusionPBX on my Debian server. Here's a brief rundown of what I expect: • Installation and configuration of FreeSWITCH and FusionPBX on a pre-existing Debian server. Operating the server is not part of your scope. • I am the only end-user. Therefore, the set-up should be tailored around my requirements and preferences. • As for functionality, I aim to utilize SIP trunking, Voice Over IP (VoIP) services, and Multi-Tenant PBX services with FreeSWITCH and FusionPBX. However, I'm interested in leveraging the complete capabilities these tools offer, so your input on additional beneficial features is welcome. To best fit this project, you should have demonstrable expertise with ...

...professional to install the Drachtio module on FreeSwitch running on my Linux OS. This will be used primarily for enhancing Voice over IP (VoIP) communication in our system. The job would entail the following tasks: - Installation of the Drachtio module on FreeSwitch - Run some tests to ensure it's functioning well and improving VoIP communication as intended To be successful in this role, you should have: - Strong experience dealing with the FreeSwitch platform - In-depth understanding and experience working with Drachtio and related modules - Strong knowledge and hands-on experience dealing with Linux OS - Good experience in enhancing VoIP communication. Keep in mind that the main target here is to install the Drachtio module on FreeSwitch and enhan...

I am looking to have FreeSwitch installed, configured with MariaDB, and integrated with FusionPBX for GUI access on a Debian Linux operating system. Key Requirements: * Install the latest version of FreeSwitch latest version. * Configure FreeSwitch with MariaDB for smooth operation * Setup FusionPBX to allow GUI access to FreeSwitch Integration part we can do later, Ideal Candidate: * Extensive experience with FreeSwitch and FusionPBX * Strong knowledge of Linux operating systems * Knowledge and experience in working with MariaDB and PostgreSQL * Forward-thinking approach, with the ability to suggest and implement beneficial features.

Edit Description In need of an experienced Freeswitch developer who can immediately tackle two key areas: configuration and deployment, as well as customization and integration. Job tasks include: - Configuration and deployment of Freeswitch - Customization according to our specific needs - Integration of Freeswitch into our existing system We missed out on specifying desired features during configuration and deployment but I welcome any suggestions from potential freelancers regarding high-availability, scalability, and security for my project. Due to the urgency of our needs, we seek freelancers who can start ASAP and have a proven track record working with Freeswitch. Mastery in system integration and customization is necessary. Ability to work quickly wit...

Hi I have a working ipsec tunnel (one host as a client and one host as a server) and I need to make sip calls from asterisk (as a client) to freeswitch (as a server) thru this tunnel. It works outside the tunnel but when I try to use the ipsec tunnel IP I can't hear audio and get call hangup after few seconds. I can provide ssh to both client and server and it's easy to 'switch' between using and not using the tunnel. Please write down the changes that you make. Also please only bid if you have experience with freeswitch. Max 100 euros. Thank you.

...expert-level training to elevate my intermediate FusionPBX and Freeswitch skills. I aim to receive comprehensive guidance covering several key areas critical to my development and the effective management of my systems. The training should provide hands-on experience and insight into the best practices for handling complex telephony systems. IDEAL CANDIDATE SHOULD POSSESS: - Proficiency in FusionPBX and Freeswitch - Strong background in configuration and setup - Expertise in system customization and development - Knowledge of performance optimization - Familiarity with security best practices for telephony systems - Experience with multi master replication, if applicable TRAINING GOALS: - Achieve advanced understanding of FusionPBX and Freeswitch - Learn effective...

I'm currently using a private VoIP provider for my Op...Opensips configuration and need help with translating specific tech prefixes. The prefixes involved are rather simple and straightforward. My goal is to ensure efficient configuration for smoother operations. For this job, you would need: - Proficiency in Opensips and dialplan configurations - Past experience with tech prefix translations - Ability to work with private VoIP providers Your role would involve: - Analyzing the current system and identifying the prefixes - Translating these prefixes within the Opensips dialplan configuration - Verifying the seamless operation of the system post-translation If you have prior experience with similar projects and are confident in your ability to make this translation, ...

I'm currently using a private VoIP provider for my Op...Opensips configuration and need help with translating specific tech prefixes. The prefixes involved are rather simple and straightforward. My goal is to ensure efficient configuration for smoother operations. For this job, you would need: - Proficiency in Opensips and dialplan configurations - Past experience with tech prefix translations - Ability to work with private VoIP providers Your role would involve: - Analyzing the current system and identifying the prefixes - Translating these prefixes within the Opensips dialplan configuration - Verifying the seamless operation of the system post-translation If you have prior experience with similar projects and are confident in your ability to make this translation, ...

Hi there, I'm in need of assistance to modify my Vicidial dialplan. The key changes I need are: - Call Routing Changes: I need to implement a system that prioritizes certain calls. However, the basis for this prioritization hasn't been determined yet and your expert opinion is highly valued. Key skills and experiences: - Extensive experience with Vicidial dialplan modifications. - Understanding of different call routing strategies. - Ability to suggest optimal call prioritization methods based on business needs. - Proficient with creating custom IVR prompts. - Familiarity with modifying the agent’s screen for better user experience. Your task will be to provide a solution to prioritize calls and implement it within the existing system. Your advice is soug...

I am at the end stage of my project and find myself in need of an experienced freelancer to integrate Google Speech-to-Text into my nearly completed freeswitch setup. Key Responsibilities: - Integration of Google Speech-to-Text into the existing project. Ideal Skills and Experience: - Extensive freeswitch knowledge. - Experience with Google Speech-to-Text integration. I am on a strict deadline and need this project to be finalized promptly. Your role is strictly on the integration of Google Speech-to-Text, relieving you from the hassle of configuring other services. I look forward to your offers.

In need of an experienced Freeswitch developer who can immediately tackle two key areas: configuration and deployment, as well as customization and integration. Job tasks include: - Configuration and deployment of Freeswitch - Customization according to our specific needs - Integration of Freeswitch into our existing system We missed out on specifying desired features during configuration and deployment but I welcome any suggestions from potential freelancers regarding high-availability, scalability, and security for my project. Due to the urgency of our needs, we seek freelancers who can start ASAP and have a proven track record working with Freeswitch. Mastery in system integration and customization is necessary. Ability to work quickly without comp...

I'm in need of an experienced FusionPBX developer who can quickly integrate vital features - call recording, call forwarding and cal...FusionPBX system - Skilled in custom API integration - Experience in GUI customization for rebranding - Ability to work under tight timeframes as this project needs to be completed ASAP Tasks: - FusionPBX Installation & system setup on Server - Integration of specified features via custom API - Brand rebranding (server setup, GUI and logo customization) Any expertise in PHP, LUA, PostgreSQL, and FreeSWITCH will be advantageous. Do note, this is an urgent project, so kindly only get in touch if you can start immediately. NOTE : PAYMENT WILL BE DONE ONLY AFTER WORK IS COMPLETED, SO IF YOU ARE IN A HURRY TO GET PAYMENT FIRST, DON'T MIN...