Asterisk pbx sbclavori

Creazione di un software che da un lato dialoghi tramite AMI (Asterisk management interface) con un centralino e dall'altro in base a delle condizioni impostabili, invii del messaggi tramite un gateway whatsappa già pronto chiamato che permette di inviare (pagando) messaggi illimitati tramite WhatsApp. inoltre questo software deve poter inviare dei messaggi anche da una lista di numerazioni che gli si fanno caricare.

Salve Alessio, sono il responsabile informatico di un'azienda di Bologna con un ufficio a Roma nel settore dello spettacolo. Dovrei provvedere alla sostituzione dei server attualmente in uso nell'ufficio di roma ed avrei bisogno di una mano in loco (via gregorio VII). Il sistema è e sarà un ubuntu server con servizi di gateway, file server samba, groupware php, posta postfix + dovecot, nginx, asterisk. Mi chiedevo se tu potessi essere interessato all'attività e magari in futuro alla manutenzione

Ciao Leandro, ho visto il tuo profilo e vorrei sapere se sei interessato ad un progetto di integrazione sw con Asterisk per fare telefonate con riscrittura del caller-id. Saluti, Leonardo

Ciao Daniele, ho visto il tuo profilo e vorrei sapere se sei interessato ad un progetto di integrazione sw con Asterisk per fare telefonate con riscrittura del caller-id. Saluti, Leonardo

Revisione di una configurazione su un centralino esistente e perfettamente funzionante

Realizzazione di una WebApp, che permetta di comunicare con PBX Asterisk: Webphone WebRTC BLF Status

Vorrei una modifica a A2billing in modo che mi permetta di vendere consulenze telefoniche. Il cliente deve inserire il suo numero di telefono e acquistare minuti di consulenza tramite paypal. Una volta completato il pagamento, il cliente chiama una numerazione geografica che lo riconosce automaticamente e lo inoltra alla coda dove risponderà un operatore. Una volta terminati i minuti dovrà abbattere la chiamata e avvisare di acquistare altro credito.

Ho la necessita di registrare un telefono cisco cp-6945 su un centralino grandstream ucm6204. il telefono ha già caricato il fw sip. Il telefono non permette la configurazione tramite web access ma solo tramite file XML

...per migliorare l'efficienza dei nostri servizi. Stiamo cercando un Consulente che ci aiuti a capire le "funzionalità avanzate" di Office 365 Business (per la parte di gestione delle email aziendali, utilizzo di OneDrive e condivisione di informazioni tra il nostro team, etc..) e l'utilizzo di Microsoft Teams (per tutta la parte di telefonia, per la parte Voip e per le funzionalità di centralino PBX, utilizzo Skype, etc..) e l'utilizzo di Microsoft Dynamics per gestire alcune attività di marketing che facciamo con del mailing (gestione dell'anagrafica aziende e anagrafica persone e gestione dei contatti - ma non ci interessa la gestione dell'offerta o progetto), ma sopratutto per gestire l'integrazione con LinkedIn recru...

...per migliorare l'efficienza dei nostri servizi. Stiamo cercando un Consulente che ci aiuti a capire le "funzionalità avanzate" di Office 365 Business (per la parte di gestione delle email aziendali, utilizzo di OneDrive e condivisione di informazioni tra il nostro team, etc..) e l'utilizzo di Microsoft Teams (per tutta la parte di telefonia, per la parte Voip e per le funzionalità di centralino PBX, utilizzo Skype, etc..) e l'utilizzo di Microsoft Dynamics per gestire alcune attività di marketing che facciamo con del mailing (gestione dell'anagrafica aziende e anagrafica persone e gestione dei contatti - ma non ci interessa la gestione dell'offerta o progetto), ma sopratutto per gestire l'integrazione con LinkedIn recru...

Per la nostra azienda disponiamo di un centralino Asterisk installato su una vps. Sto cercando una persona che possa intervenire ogni tanto per eventuali modifiche alle funzioni del centralino. Vorrei stabilire una tariffa oraria per gli interventi che potranno essere fatti da remoto.

Montar una centralita ASTERISK en una oficina.

Buonasera, dobbiamo risolvere il problema di interfacciare una nostra applicazione gestionale Web con Asterisk, in modo che, in primo luogo, si possa chiamare un numero di telefono dall'applicazione (in secondo luogo innescare un popup all'arrivo di una telefonata). E' disponibile per una consulenza? In caso, il mio indirizzo di posta è at

Buongiorno, ho un crm scritto in vb.net che si interfaccia con asterisk/vicidial. deve essere perfezionato e corretto.

Da creare un software di calendarizzazione delle chiamate. Dove in un panello, possiamo fare una programmazione delle chiamate, e poi in automatico il software fa le chiamate. Il software deve ...automatico il software fa le chiamate. Il software deve servire a far funzionare un Gateway gsm voip, all'interno del device vi vengono inserite delle sim card, il programma deve far le chiamate in automatico, previa programmazione all ora e al numero e dalla sim (lo slot dove è collocata) pre stabilito dalla programmazione. Molto probabilmente si dovrà realizzare un programma che utilizzando un PBX voip tipo asterix utilizzi i modem tramite lo strato di rete ethernet fornito e tramite una programmazione specifica del centralino. Allego i file manuali del nostr...

Sto cercando un esperto asterisk che sia disponibile ad alcuni lavori di manutenzione e modifica del nostro piccolo centralino asterisk.

Cerco una persona per creare un programma per la gestione di una rubrica (excel o access) che mi crei un file XML con particolari caratteristiche per essere letto dal PBX Yealink. Il software deve rimanere per uso interno quindi non è necessario un particolare design o altro. Deve essere semplice, funzionale, modificabile, facile nella lettura e nella scrittura. Il progetto è relativamente piccolo in quanto la sintassi è già definita dal PBX, resta soltanto da creare la macro per la creazione del file XML da usare in remoto o locale. Il budget quindi non sarà elevato, progetti cari saranno scartati.

Good morning, my name is Fabiano and I need your help. I need to install Asterisk + FreePBX + FO2 panel on a vps. distribution choice centos 5.9 32 or 64bit centos 6.6 32 or 64bit centos 7 64bit centos 7.1 64bit Fabiano

Cisco PBX 2921 , troubleshooting.

Hello , we ARE LOOKING FOR REAL PRO TO SETUP FOR US ASTERISK WITH SIP AUTO DIALER. pLEASE HELP HERE.

Ho necessita' di configurare centealino voip con programma free pbx + Software virtual weather exchange

...europee/US ( inglese e italiano in particolare) e con un suono dolce - sarà facile da ricordare - non dovrà evocare per assonanza concetti impropri/negativi nelle principali lingue straniere (italiano e inglese in particolare) - sarà distintivo rispetto alla concorrenza (non dovrà confondersi rispetti ai nomi delle aziende di comunicazioni multicanale, telecomunicazioni, call center, produttori PBX e servizi web presenti sul mercato) - dovrà essere un nome coinvolgente, stimolante, emozionante. Di seguito i requisiti per il dominio internet della nostra BU: - dovrà corrispondente al nuovo naming della BU - dovrà avere estensione .com e .it - dovrà essere un dominio. libero sul web e da problemi legali, non regi...

...Integrations:** - **Messaging Platforms:** WhatsApp, Facebook Messenger, email, and SMS. Additionally, PBX system integration is required to bridge our voice call capabilities with digital communication channels. - **Features:** Essential features we're looking for include: - **File Sharing:** Ability to share documents, images, and videos across all channels. - **Chat History:** Seamless access to previous conversations for context and continuity in customer support. - **Chatbot Integration:** Capability to integrate AI chatbots for automated responses and assistance. **Ideal Skills and Experience:** - Experience with API integration for WhatsApp, Facebook Messenger, email, SMS, and PBX systems. - Proficient in developing chat platforms with functiona...

Estamos à procura de um especialista em Issabel e Asterisk para implementar um sistema de Resposta Vocal Interativa (IVR) na nossa plataforma de servidor Issabel 4.0.0-6. O objetivo deste sistema é realizar entrevistas telefónicas automatizadas onde os respondentes possam responder a perguntas pré-gravadas utilizando o teclado do seu telefone. O sistema deve ser capaz de carregar uma base de contactos para realizar automaticamente as chamadas telefónicas e capturar estas respostas numa base de dados para análise subsequente. Além disso, o sistema IVR deve incluir a funcionalidade de conversão de texto em voz (TTS) para facilitar a criação e atualização de prompts de voz. No entanto, também d...

...solution, I'm seeking guidance from a skilled freelancer experienced with both Raspberry Pi and Asterisk server setup. Key Project Details: - I do not require a full-fledged PBX setup, just a basic VoIP service facilitated through my Raspberry Pi. - The primary feature I'm interested in implementing is voice calling. Required Skills: - Proficiency in configuring an Asterisk server on Raspberry Pi. - Strong understanding of VoIP and related protocols. - Ability to guide and explain the setup process clearly. Your Role: I've tried to set it up but no voice can be heard on the other end. So this task is mainly a trouble shooting job. Your primary role will be to walk me through the setup of Asterisk on my Raspberry Pi, ensuring proper configur...

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.

...persona con experiencia configurando la libreria de Javascript SIPML5 y asterisk. Tenemos todo instalado y configurado. El softphone web se registra al asterisk, emite y recibe llamadas, pero cuando se atiende la llamada no transmite el audio. El servidor tiene instalado una VPN. Cuando el usuario se conecta a la VPN, entonces funciona el audio de la comunicación pero cuando no se conecta a la VPN entonces vuelve el problema del audio. Se requiere que el softphone funcione sin VPN, solo por internet. Version asterisk 18 OS: Ubuntu Server 14 +++++++++++ A person with experience configuring the SIPML5 and Asterisk Javascript library is required. We have everything installed and configured. The web softphone registers to Asterisk, sends and recei...

I'm in need of an expert who can help install and set up an Alcatel PBX system for a medium-sized enterprise. The project also involves integrating this PBX system with VoIP services. This will be remote project, our team will give you access from the site. Key Responsibilities: - Configuration of Alcatel PBX system OXO408 - Integration of the PBX system with VoIP services along with call recording. - Integration with existing Cisco IP Telephony Ideal skills for this job include: - Profound understanding of Alcatel PBX systems. - Experience with VoIP services integration. - Prior work on medium-sized enterprise deployments. Your expertise in these areas will help us to establish a reliable communication system for our business, ensuring seamless...

Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error. [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descr...

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descri...

Hi, I have one sample codes which work as voip client . it should run on ARM32 . and I have one rockchip rv1106 linux SBC (I can add memory and emmc size following your advise). this rv1106 has its SDK, I installed on virtual unbuntu 20.04,and buid SDK and its sample application codes ok, but when I integrate the voip sample code into sample application codes ,it has link error about GLIBC,etc ,similar outputs as this url: I tried to fix it myself ,and asked for help ,still not working .I guess rv1106 cross compiler tool shipped by chip provider is uclib (arm-rockchip830-linux-uclibcgnueabihf ) ,but the voip sample codes need glib. so I am thinking maybe we should use buildroot to build a new linux image

As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.

Good day, everyone. We are seeking a VoIP system developer to create the following features from scratch for our system: We are VoIP ...sign-up option. 2. The portal should allow customers to operate on a pay-as-you-go system, enabling them to recharge or pay their bills directly from the portal using various payment methods such as cards, bank transfers, or cryptocurrencies. 3. We aim to integrate a dialer system within our customer portal, allowing clients to create webphones for their agents/users and utilize other dialer or PBX features. We are open to demonstrating our current working system to provide a better understanding of our requirements. We are seeking a scalable solution, and if anyone has innovative ideas, we are more than willing to explore and collaborate on thos...

Requiero ayuda con unas horas de soporte para aclarar algunas dudas sobre una integracion entre una central Grandstream y Opera PMS.

...1: Contará con un Home Gateway. Cada uno de ellos tiene que estar registrados al servidor remoto de la empresa y conectados a su Home Gateway. Se tiene que poder administrar a través del HomeGateway. Todos los dispositivos tienen que estar conectados de manera inalámbrica. Los dispositivos son: Cafetera. Parlantes Puerta. Lámpara. Ventana. Adicionalmente, la casa tiene que contar con un SBC que esté configurado con: Una cámara web Un sensor de Movimiento. Una puerta Una Ventana Una Sirena. Un led RGB. Cada uno de estos componentes tiene que interactuar y enviar un correo al departamento de seguridad de la empresa. tienen que entregar un video explicativo en el que participará cada uno de los miembros del grupo y, dependien...

I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50

We are looking for an engineer proficient with Raspberry Pi, as we are in need of developing a VoIP PBX system on a Raspberry Pi 3 Model B+ . The end goal includes the integration of specific features into the system such as: - Call Recording - Voicemail to Email - Conference Bridging - Additional bespoke features Your expertise should include not only Raspberry Pi but also Asterisk and RASPBX/FreePBX. We are aiming for a robust, stable, and user-friendly system with custom features tailored primarily to business needs. The successful contractor will be required to develop the system on his own Raspberry Pi and submit an IMG file for loading onto other Raspberry Pis. The successful contractor will be working with our software developers so bids from individual contractors...

Necesito ayuda con la configuración de mi Asterisk, ya que no soy capaz de recibir el DTMF, con la opción que selecciona el usuario, tras escuchar la locución de la IVR. He intentado transcodificar la señal, y aplicar diferentes configuraciones, y codecs sin éxito.

I'm urgently looking for a skilled professional to quickly handle an IVR-related task for me: - Make IVR calls to 120 different cell phone numbers delivering specific information. Exp...IVR-related task for me: - Make IVR calls to 120 different cell phone numbers delivering specific information. Experience in both script writing and call routing configuration is appreciated, although not mandatory. The swift initiation and completion of calls is paramount to this project. Therefore, the ideal freelancer will demonstrate adeptness in utilizing any form of IVR system, be it traditional PBX, cloud-based, or hybrid system. Please note that this project is time-sensitive and needs to be started and finished ASAP. Your adaptability and readiness to start immediately will be h...

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

...BIM/MEP Engineer, proficient in utilizing Revit, AutoCAD, and Navisworks. The selected candidate's scope of tasks will encompass: - Assisting with comprehensive 3D modeling - Conducting effective clash detection - Providing technical support - MV/LV Networks , (cable routing, medium voltage & low voltage switch gears, transformer loading). - load calculation and load estimation according to SBC code - power single line diagram cable sizing, voltage drop, and short circuit calculation. Sockets, power outlets. - Load schedule. - street and exterior lighting / Interior Lighting design. - Grounding systems. - ICT - Sound system, - Cctv system, Ideally, the freelancer should have experience not only in design but in design implementation as well. A proven track record o...

I am urgently seeking an experienced telephony and data processing specialist to configure my Grandstream UCM6302A with Asterisk. The core functionality required includes receiving calls, playing a welcome message, meanwhile working with Caller ID and Web API to determine where to forward the call. When a call comes in, • first a welcome message is played () • in the meantime the caller ID will be sent to web API preferably POST, but get can be if POST is not possible () •The API will respond json array: - {forward_to: 33356853 } - Forward the call to 33356853 - {forward_to:0 } - Play message and terminate the call • If forwarded call is not answered by Agent in three rings, another call to API will

...experienced freelancer to set up a Voice over Internet Protocol (VOIP) system for a yet to be determined number of lines. As no preference for current phone system was specified, I am open to recommendations and would ideally like someone who has experience with both traditional landlines and hosted VOIP systems as well as virtual PBX systems. Ideal Skills and Experience: - Experience in VOIP system installations - Knowledge of Traditional landlines, Virtual PBX, Hosted VOIP systems - Ability to recommend suitable solutions depending on project needs Please include in your application: - Your past work related to VOIP systems installations - Specific experience you have in relation to this project's needs - A detailed project proposal outlining how you will approac...

...am looking to incorporate AI features into my call center system, specifically Vicidial and Asterisk. As these platforms form the core of our operation, it is essential that any alterations enhance our outlay without disrupting the existing structure. Key Aspects of the Project: - AI Implementation: Even though I haven't specified the exact AI features to integrate, I'm interested in potential focus areas such as speech recognition and transcription, natural language processing, or sentiment analysis. Proposals that offer comprehensive strategies addressing these or other AI fields will be highly considered. - Dual Integration: The AI features must be incorporated into both Vicidial and Asterisk, aligning and harmonizing their performances. - Efficiency Goal: ...

i need someone to teach me how to upgrade firmware of cisco 7821-k9 to make it use sip protocol to hook it up on asterisk pbx More details: Which specific features do you require for your Cisco 7821-K9 SIP protocol project? upgrade firmware , connect it to asterisk as sip extension Which version of firmware would you like to upgrade your Cisco 7821-K9 SIP protocol to? Latest firmware version What functions do you require for the SIP extension with your Asterisk system? Call Recording, Call Transfer, Multi-Line Functionality thank you very much

I'm in need of a skilled voice-over artist to inject warmth and friendliness into my PBX system. Meeting the guidelines below is crucial: - Provide a voice-over in a warm, friendly, and conversational style - The recording should be in English preferable US Englisch and range between 1 to 5 minutes - it's 2 scripts but each for 4 different company names. I'm fine if you cut the names so you do not need to record the scripts each 4 times. - Ideally, the voice-over artist should have a middle-aged vocal range I'm looking forward to hearing your warm and engaging voices.

I have a Twilio account with sip trunking set up, and I've install Asterisk on Arch Linux, I've attempted to set up the config but have not been able to. I'm looking for someone to set up a basic config where I can send and receive phone calls. The details don't matter, I just want to get it working so I can adjust it once the simplest config is working. If interested please bid the amount you are able to do this for, and include the word "briefcase" in your bid so I know you've read the description and can complete the project for the amount bid.

I'm seeking an expert...integrating Zoho CRM and 3CX for fewer than 10 users. The key goal of this integration is to set up 3CX telephony within Zoho CRM. I tried but failed and don't know what I'm doing wrong. Key responsibilities: - Calls are to be looked at via the Zoho CRM telephony function and lookup contacts to show who is calling. Looking for: 1. Jump on a call via Zoom 2. Help me read: 3. Via Zoom guide me or provide support on what I'm setting up wrong. 4. Test solution or help me contact support if needed. This project offers the opportunity to bring significant improvements to our CRM and telephony capabilities. I look forward to engaging with seasoned professionals who can efficiently and effectively execute this project.

...featured communication app for both iOS and Android. This app will connect with my existing Asterisk server through APIs. Key Features Include: - User creation - Real-time balance display - Call-making functionality - Fully integrated payment gateway - Text messaging - SIP voice calls (Not video calls, just normal SIP calls) Necessary Skills and Experience: - Proficient in iOS and Android app development - Proven experience with PortSIP SDK - Familiarity with Asterisk and its relevant APIs - Skills in developing chat features, specifically text messaging and voice calls, within an app - Experience in implementing a payment gateway in an app Please note, I have access to and can provide the necessary Asterisk API documentation. Ideally, you are able to show me a s...