Avaya asterisk sip trunklavori
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.
ho un centralino yeastar e ho un trunk con voippro, quando effettuo una chiamata da un interno verso l estreno non sento la voce del chiamato ma il chiamato sente tutto. Se chiamo gli interni funziona tutto perfettamente. se ricevo le chiamate dall estreno funziona tutto perfettamente
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 Raffaele, ho visto il tuo profilo e da quanto letto suppongo ci siano i presupposti per sottoporti un lavoro, anche con la semplificazione della lingua, che dovrebbe essere per noi un concept da discutere col cliente per capire se renderlo produttivo o meno entro 1 mese. Le tecnologie che verranno toccate saranno principalmente webrtc, js e sip.
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
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.
Realizzazione script di configurazione dispositivo PATTON SN-DTA/1BIS2V/EUI per collegamento con centralino ISDN SAMSUNG iDCS 100. Il PATTON deve essere configurato con due TRUNK VoIP Eutelia (attuale Cloud Italia). Implementation Script Configuration to connect a PATTON SN-DTA/1BIS2V/EUI to SAMSUNG iDCS 100 ISDN PABX. The PATTON must be configured with TRUNK VoIP Eutelia (current Cloud Italia).
Realizzazione script di configurazione dispositivo PATTON SN-DTA/1BIS2V/EUI per collegamento con centralino ISDN SAMSUNG iDCS 100. Il PATTON deve essere configurato con due TRUNK VoIP Eutelia (attuale Cloud Italia).
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.
i need voip expert more detail in pmb .....................................................................................................
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
Diffusione di musica via multicast in formato ulaw a vari apparati SIP " SNOM PA 1 utilizzando come sorgente l'ingresso MIC di un PC, tramite VLC o altro software. in questo modo collegando l'uscita AUX di uno stereo analogico all'ingresso MICROFONO del pc si ottiene una diffusione di musica agli SNOM PA 1 Possibilità di diffondere anche flusso di radio internet ____ Distribution of music via multicast format ulaw to various apparatuses SIP "SNOM PA 1 using as source the MIC input of a PC, via VLC or other software. in this way i need to send music from ANALOG RADIO to SNOM PA 1 Possibility of spreading also stream Internet radio
Diffusione di musica via multicast in formato ulaw a vari apparati SIP " SNOM PA 1 utilizzando come sorgente l'ingresso MIC di un PC, tramite VLC o altro software. in questo modo collegando l'uscita AUX di uno stereo analogico all'ingresso MICROFONO del pc si ottiene una diffusione di musica agli SNOM PA 1 Possibilità di diffondere anche flusso di radio internet ____ Distribution of music via multicast format ulaw to various apparatuses SIP "SNOM PA 1 using as source the MIC input of a PC, via VLC or other software. in this way i need to send music from ANALOG RADIO to SNOM PA 1 Possibility of spreading also stream Internet radio __
Sono alla ricerca di firmwarista in grado di programmare il microprocessore Texas TMS320DM368, o altro similare, con le seguenti funzioni: Connessione parallela a CCD ( telecamera) Codifica video H264 NV12 BP/MP/HP Codifica audio: G.711?/a, G.722, G.726-32, G.729, (RFC2833, SIP INFO) Trasmissione video e audio full duplex su IP tramite server SIP Sinterizzazione brevi messaggi audio Sovraimpressione testo su immagine video Altre piccole funzioni
Hello , we ARE LOOKING FOR REAL PRO TO SETUP FOR US ASTERISK WITH SIP AUTO DIALER. pLEASE HELP HERE.
I'm looking for someone to help me set up a SIP trunk on my FreePBX phone system. Key Requirements: - Configuration of SIP trunking: I need you to assist in the setup of a SIP trunk on my FreePBX system. Your expertise in this area is crucial. - Connecting to a specific SIP trunk provider: I have a specific SIP trunk provider in mind, and I need you to ensure that the FreePBX system is correctly connected to this provider. Ideal Skills and Experience: - Proven experience with FreePBX phone systems and SIP trunking: I'm looking for someone who has successfully configured SIP trunking on FreePBX systems before. Please provide examples of similar work. - Familiarity with various SIP trunking pro...
...highly-skilled Asterisk VoIP specialist for a unique project. I require expertise in Asterisk integration, VoIP configuration, and particularly in IVR implementation. KEY REQUIREMENTS: - Asterisk Integration: Full integration set up and testing needs to be completed. - VoIP Configuration: This includes configuring phone lines, extensions, and all essential VoIP functions. - IVR Implementation: The main purpose of this project is the implementation of automated IVR menus. This needs to be user-friendly and functional. The system should be capable of handling less than 50 concurrent calls. IDEAL CANDIDATE: The successful freelancer must have outstanding experience in all mentioned areas with a focus on IVR implementation. Any proven record in developing low-volum...
I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.
The objective of this project involves the implementation and modification of an open SIP protocol to enable data synchronization between mobile devices and computers. The successful freelancer should possess: - A comprehensive understanding and experience with SIP protocols - Ability to implement and customize existing protocols for specific needs - Proficient in facilitating cross-device connections, specifically between mobile devices and computers - Familiarity with data synchronization methods and applications By the end of this project, we aim to have seamless data synchronization between the devices in question. As such, creative problem-solving skills and a keen attention to detail will be invaluable.
I am setting up a call center and need assistance activating SIP via VPN on Mikrotik OS. The SIP server for the call center will be accessed through a site-to-site VPN using IPsec. Key requirements include: - Configuring Mikrotik OS for SIP calls - Establishing secure site-to-site VPN using IPsec - Ensuring the SIP server is accessible through the VPN - Preparing the system to handle more than 50 concurrent calls at peak times Experience and Skills: - Expertise in Mikrotik OS - Proficiency in SIP server configuration - Strong understanding of VPNs and IPsec - Prior experience with call center setups would be advantageous - Ability to design systems that can handle high call volumes is essential Please include any relevant experience or examples of sim...
I am looking for an experienced professional to install a SIP trunk server on a Linux system. As part of the project, you should also set up access to the live voice stream with the primary purpose of real-time transcription. Key Requirements: - Install and configure a SIP trunk server on a Linux system. - Enable access to a live voice stream for real-time transcription. Ideal Candidate: - Proficient in Linux server administration. - Experienced in SIP trunk server installation. - Familiar with real-time transcription processes. - Skilled in setting up live voice stream access. - Strong problem-solving and troubleshooting abilities.
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.
General Event Assistants wanted for a fun, science-themed event. The Lit Lab (@litlablondon) is the UK’s first and only ‘Science and Sip’ experience. In this exclusive, immersive pop-up event, guests will be transformed into scientists to create exciting drinks, conduct science experiments, and compete in challenges against other scientists. Responsibilities: - Act as general event staff for monthly events and some ad-hoc events in London. - Event dates and times TBC. These will be shared at least 1 month in advance where possible, at which time you can accept or decline within 1 week. - Temporary / day work: 4-5 hours work. This is a pop-up event that changes locations. - Setup and clean-up: Moving and arranging chairs, tables, equipment and storage boxes e...
...ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SI...
...branding guidelines. Test modifications thoroughly to ensure stability and functionality. Collaborate with our internal team to gather requirements and feedback. Provide documentation and support for the customized software. Requirements: Strong proficiency in Visual C++. Experience with software customization, particularly with open-source projects. Familiarity with VoIP technologies and SIP protocols is a plus. Excellent problem-solving skills and attention to detail. Ability to work independently and manage time effectively. Project Details: This is a freelance, remote position. Duration: Approximately 1-3 months, depending on the project scope and requirements. Payment will be negotiated based on experience and the specifics of the agreed-upon deliverables. ...
I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this...
I am looking for an expert who can help me set up a SIP trunk server using Twilio as the SIP provider. This server will be used for both voice calling and SMS messaging. Key Responsibilities: - Establishing a SIP trunk server with configurations optimized for Twilio - Implementing call management features like Call Recording, Call Forwarding, and Interactive Voice Response (IVR) - Ensuring seamless integration of both voice and SMS capabilities - Providing recommendations for server security and performance optimization Ideal Candidate: - Extensive experience with SIP trunking, particularly with Twilio - Proficiency in configuring and customizing SIP trunk servers - Proven track record in implementing call management features such ...
...ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SI...
Seeking an experienced Asterisk developer to optimize the call quality of our existing VoIP system within our business operations. The ideal candidate must: - Possess 3+ years of experience in the domain - Be proficient with programming languages, VoIP, Asterisk interfaces (ARI, AMI, AGI), SIP configuration, API integration, and webhooks - Have robust problem-solving skills to provide innovative solutions Key responsibilities will include scrutinizing our present setup, identifying weaknesses, and implementing improvements to enhance call quality. A solid understanding of business requirements is vital to ensure the VoIP system is modified to suit our operational needs. Interested candidates, please email your resumes and cover letters.
I'm looking for an expert in VoIP/SIP setup, with experience in setting up contact centers, to help us with our inbound customer support setup in the US. The main requirements for this project are: - Setting up a VoIP/SIP system for inbound customer support for Indian callers - Managing this setup to accommodate more than 50 agents - Integrating the system with CRM software and a Ticketing system You should have: - Proven experience with similar projects - Expertise in VoIP/SIP setup and managing large contact center environments - Knowledge of CRM software and Ticketing system integration Your suggestions and recommendations are welcome.
...design for a barn workshop. This barn should have a large open studio area with lean-too on both sides. Mortis and tenon joints and dovetail mortis and tenon joints to be used every that a structural connection is required. Frame inside walls if studio space will be sheathed with 2x4 stud walls insulated with batt insulation with shiplap on lean top side and studio side. Roof will be capped with SIP panels. Roof will be metal and outside walls will all be metal siding. Key Requirements: - 3D Model and Design to be done in Revit or other approved software. - Per Michigan building code and ASCE 7 for design loading for SE Michigan. - All design files to be provided including 3D model, hand calcs, and pdf plan sheets with overall plan, elevation views, and individual member detai...
Configurar router y dispositivo para registrar ina linea SIP de O2
...framing details, mechanical, electrical, or plumbing plans, window or door schedules, a site plan, foundation plan, gutter or drainage plan, or 3D renderings. Since the exact plans you produce will not be inspected or built, such details are not important. The plans must only serve as a realistic and potentially buildable example of this kind of project, so I can get and compare estimates from SIP (Structural Insulated Panel) providers and builders about potential costs of constructing such a the building envelope. Therefore, while you are not specifying panels and construction methods, it would be helpful if you have an understanding of SIPs as used in US/Canada residential construction. You would only be hired after a video meeting to determine your ability to understand the...
We need to customize linphone's windows/macOS/iOS/android apps with regular PBX features, sms,sip Presence,BLF,etc. Needs to be customize with our company logo/colors. Change colors and logos to match our theme. Sleek and professional design. Login screen to accept username and password or scan QR code Incoming and outgoing calls. Call Transfer (Blind and assisted transfer) Hold Call Transfer Call Attended Transfer Call End Call Conference Call (merge the calls) Video calls Presence and IM features Call history Smart contact list, with address book synchronization for smartphones Audio/video conference calls and scheduled meetings Call transfer and multi-call management (pause and resume) One-to-one and multi-participant conversations Intuitive message de...
My SIP Trunking system is constantly preventing me from receiving inbound calls. It's a constant issue that requires urgent attention. Ideal candidates will possess: - Extensive experience with SIP Trunking - Knowledge in diagnosing and resolving call reception problems - Proficiency in troubleshooting technical issues constantly, not sporadically.
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. ...
...love, Returns and Replace policy - We need to have a collections page, that lists all products under that collection - The rest of the marketplace will be pretty standard - Our header and menu navbar are going to be separate for all pages. The header is supposed to be a branch of a tree with the sitelogo on it in the middle. - On a desktop view we want the left side of the screen to have a tree trunk run along on the left till the footer And leaves on the right. - Footer content should include: Column 1: Small kalataru logo and under it social media links Column 2: About us Contact us Careers Kalataru Magazine Column 3: FAQs Returns and exchanges Shipping and Handling Terms of Service Privacy Policy Column 4: Sign up to our email Newsletter (Email ID text box with validation) (...
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.
I'm looking for an experienced freelancer to help me set up the SIP locally to connect incoming calls to an AI script I have. This setup has to be done on VICIdial using the QuestBlue SIP. The main objective of this project is to establish a call response system. As someone with a pre-built AI script, I need you to link it to the incoming calls on the VICIdial platform. Your expertise will include, but not be limited to: - Proficiency in VICIdial: You must have previous experience working with this platform. - Familiarity with SIP set-ups: QuestBlue SIP experience would also be a bonus. - AI Integration: Able to connect incoming calls to an AI script. Please ensure these skills align with your capabilities before bidding on this project.
