Asterisk huawei freepbxlavori

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

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

...pubblico fastweb e la ge2 con ip privato Nell'ufficio 2 abbiamo vodafone configurato con indirizzo ip pubblico e un router HPE MSR2003 configurato sulla ge0 con ip pubblico e la ge1 con ip privato sulla porta ge2 vogliamo configurare la parte che attualmente non abbiamo collegato perchè ha gli stessi indirizzi dell'ufficio 1 nellufficio 3 abbiamo il router vodafone huawei configurato con ip privato Non riusciamo a comunicare con tutti i dispositivi nelle varie lan con i router hp connessi quindi sicuramente c'è qualche problema di configurazione sui router hp. I router hanno il sw Comware version 7.1.064...

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 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.

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

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

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

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...

I am looking for a competent professional with experience in Freepbx and Avaya IP integration. The primary objectives of the project are call routing, call forwarding, and establishing a unified voicemail. What I am looking for: - Experience with Freepbx and Avaya IP office 500 V2 systems - Proven track record with telecommunication technology - Capability to integrate functionalities even with gaps in specific call routing or forwarding options Please note that while some specifics on the call routing and forwarding options are not outlined, I expect freelancers to provide suggestions based on their experience and expertise. I look forward to working with an individual who can make these systems work smoothly together.

...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 configuration f...

I'm in need of a highly qualified and certified network and security engineer. Ideally, you should have certifications from Cisco, F5, Symantec, HP, HPE, Dell, Huawei, Avaya, Fortinet, and Juniper. Key Responsibilities and Tasks: - Network design and configuration - Firewall management - Network troubleshooting - Telecommunication system setup - Network security audits - Hardware installation and maintenance - System integration - Vendor selection and management - Providing ongoing support Your background should demonstrate extensive experience with all the above functions. I'm particularly interested in freelancers who have successfully handled similar tasks in the past. Your success will hinge on your ability to improve our overall network and security infrastructure. ...

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...

Dados complementares para gravação das 130 frases: -> duração de gravação em torno de 20 minutos -> Assine apenas seu nome e ultimo sobrenome -> Gravação em telefones celulares de última geração, com no maximo 3 anos de idade: Apple, Samsung e Huawei por exemplo. -> Sobre o Ambiente de gravação: muito silencioso, sem nenhum eco (o som pode ser absorvido com a adição de edredons e travesseiros) -> É necessário que haja uma pausa (um segundo mais ou menos) entre a Alexa e a palavra de comando -> A velocidade geral de fala é um pouco mais rápida -> A distância entre o microfone a sua boca deve ser cerca de 20 cm -> Volume de...

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.

...developed for delivering OTP messages through a web interface. Short codes will need to be delivered through web panel to customers. Text message delivery for multiple services used in accounts creation like facebook, instagram, gmail, whatsapp and other services.. This means we will be using physical sim cards, NOT virtual numbers. Web backend will need to communicate with physical hardware: (Moden huawei e303, 3131 FOR EXAMPLE) The system has to be the same as sms-activate, sms hub, sms pva, please if you have this experience I am searching for coders experienced is USSD/AT commands. Heavy experience with GSM/CDMA cellular networks and sms gateway integration. Look at the websites I've posted above, put together a compatible hardware list for the web frontend&backen...

Dados complementares para gravação das 130 frases: -> duração de gravação em torno de 20 minutos -> Assine apenas seu nome e ultimo sobrenome -> Gravação em telefones celulares de última geração, com no maximo 3 anos de idade: Apple, Samsung e Huawei por exemplo. -> Sobre o Ambiente de gravação: muito silencioso, sem nenhum eco (o som pode ser absorvido com a adição de edredons e travesseiros) -> É necessário que haja uma pausa (um segundo mais ou menos) entre a Alexa e a palavra de comando -> A velocidade geral de fala é um pouco mais rápida -> A distância entre o microfone a sua boca deve ser cerca de 20 cm -> Volume de...

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...

I'm looking for a FreePBX expert to help configure the Sticky Agent feature on my system. Key Requirements: - Configure Sticky Agent - Route call to the same agent who answered previously. - If that agent are unavailable, then route to his VM. Call Flow: - Inbound Route - Queue - Static Agent Ideal Skills and Experience: - Expertise in FreePBX, specifically version - Only apply if you have done this previously. My server is live, hence no experimenting. Budget $50

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.

I need an expert who can modify my wife's Huawei Chinese phone to be fully compatible with American apps and servers. This is crucial as the phone was an expensive purchase and we can't afford for it to be useless in America. Key Requirements: - Experience in programming Huawei phones - Strong knowledge of the Android operating system - Ability to modify apps and servers to be compatible with the Huawei platform. The successful freelancer should have a track record of similar projects and provide a portfolio or examples of past work. Please do not hesitate to reach out for further information.

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...

Task: Need to send a GSM command string ( USSD or SIM Tool Kit commands ) to request a pre-paid mobile credit send, credit recall, and balance check. These commands will be triggered by a web server, and communicate with the web server and log all events. Options: - Create stand alone software that will communicate with the web server. - Us...These commands will be triggered by a web server, and communicate with the web server and log all events. Options: - Create stand alone software that will communicate with the web server. - Use the Diafaan software or similar, and customize with it’s C++ / .Net scripting connector - Use other pre existing software Current Hardware: Windows Machine with remote access Port modem , port Huawei modem 3 SIM cards from L...

I require an experienced Huawei BSC 6900 expert to help with the configuration and optimization of our BSC 6900 system. Key tasks will include: - Focusing on efficient and optimal base station control. - Overseeing resource management to ensure proper utilization and efficiency. - Specifically dealing with ongoing connectivity issues. This project is a fit for you if: - You have prior experience with Huawei BSC 6900 system running R09 - You have resolved connectivity issues in the past successfully. - Your expertise lies in network configuration and optimization. Your goal will be to optimize the BSC 6900 system to ensure enhanced connectivity, resource management and base station control.

...lower in crowded cities. Therefore, we need to change the backhaul connection of our gateways to 4G mobile Internet. There are two approaches to this: 1) using an external 4G modem that has an Ethernet connection (), 2) connect a 4G USB stick to the USB connector of the Kerlink Wirnet Station, like the Huawei K5150 (), or the Huawei K5160 (). As we also have some of the aforementioned 4G USB modems, what we are looking for is a solution for the 2) option, that is the integration of the K5150 or the K5160 to the Kerlink Wirnet OS. The solution should be able to handle both public and private APN-s

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.

Description: We are seeking a skilled developer to create a Visual Plan application for FreePBX 15 or FreePBX17 using Node.js or PHP. The application will be based on the Open Source project of FreePBX 13 available on GitHub (), but the candidate is free to decide whether to start from scratch or use an alternative library such as Draw2D (). The main purpose of this Visual Plan is to visualize and modify the flow of the phone system. Requirements: Proficiency in Node.js or PHP, depending on the chosen application Experience with FreePBX and its configuration Ability to work with graphic libraries like Draw2D Understanding and implementation of functional requirements for the Visual Plan Ability to

I'm facing a critical issue with my FreePBX server. To be specific, I'm unable to make outgoing calls despite not having made any recent changes or updates to the server. Furthermore, while attempting these calls, I encounter a 'No response from server' error message. This problem needs an experienced and knowledgeable freelancer to identify and rectify the issues. Key requirements for this task: • Expertise in FreePBX server maintenance and troubleshooting • Ability to work promptly, as this issue is significantly affecting my operations • Solid understanding of host networking and telecommunication systems • Experience in resolving 'No response from server' errors or related issues. Resolving this issue will restore the com...

Integrated Platform for Website, Control Panel, Content Management, and Mobile Apps (Android, iOS, Huawei) Overview: The project aims to develop a comprehensive platform that includes a website, a control panel, a content management system (CMS), and mobile applications for Android, iOS, and Huawei devices. This integrated platform will provide seamless access to content, management tools, and user interfaces across different devices, ensuring a cohesive user experience. Components: Website: Develop a responsive website with user-friendly interfaces. Implement modern design principles for aesthetic appeal and usability. Include essential features such as user registration, login, search functionality, and navigation menus. Ensure compatibility across various web browsers ...

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

...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 would like to launch my application on the stores apple store, google store, huawei. And maintain my servers on azure or AWS with local database on my device only .net developer with experience of azure and AWS is required The app is on the stores The product is ready and deployed

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.

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Hi, i need help in setting up a home server. I have a Huawei 5G CPE Pro 2 router connecting my house to the internet. I'm trying to setup a small home server. I have setup port forwarding on the router on multiple ports to the server private ip. the i'm trying to connect the router public ip to resolve to that server. I need help in doing that. Only people with experience in this subject will be accepted

...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...

...conducting comprehensive testing of the installation and purchase processes of applications on various models of Chinese devices with pre-installed native markets (not Google Play). My company needs service who can test the installation and purchase processes of GOMINING application on various Chinese device models IN South Asian countries. The test involves pre-installed native markets, specifically Huawei App Market or any devices with no Google Play (Chinese Android Phones) Here's what we are looking for: - Someone who is fluent in Chinese to help with testing on up to 20 phone models sourced from three or more Southeast Asian countries (Malaysia, Singapore, Thailand, Vietnam, Taiwan, Hong-Kong etc.). - We need to see the usability of the app on Chinese android phone w...

I am currently experiencing an issue with the FreePBX phone system that I use, specifically, I'm unable to consistently receive calls on all my soft virtual extensions or to my welcome greeting. Voicemail functionality is also part of this setup. Here's what I need: - Evaluation and Diagnosis: Investigate why call reception works only some of the time across all extensions. - Troubleshooting and Repair: Identify the problem and implement a fix to ensure consistent and reliable call reception. Ideal Skills Required: - Extensive experience with FreePBX Phone Systems, particularly concerning soft virtual extensions and voicemail functionalities. - Strong understanding of VoIP systems, and hands-on experience in troubleshooting call reception issues. It is cr...

I need assistance setting up a new SIP Trunk on my FreePBX 14 system. Here's what you will need: • Experience with FreePBX 14 • Familiarity with the setup of SIP Trunks • Knowledge of goip local voice gateway Tasks to be accomplished: • Set up a new SIP Trunk on my FreePBX 14 • Ensure the setup is correctly done using my service provider, goip local voice gateway. Only those with the prescribed skills and experience should bid, as I need this done quickly and accurately. Thanks!