Nelly codec sip rtp işler

PBX telefon centrali uzmanı Genel Nitelikler - İleri Düzey PBX Sistemleri ve Voip Gatewayler hakkında bilgi sahibi. - İleri Düzey FusionPBX, Freeswitch, 3CX telefon centrallerinde uzman. - Sip telefonları provisioning yapabilen. - Tercihen linux scriptleri yazabilen. Atomasyon yapmaya meraklı. - Network bilgisine sahip (TCP/IP, LAN, WAN, VPN) - VoIP proje ve uygulamaları konusunda en az 3 yıl deneyimi olan - WAN uygulamaları ve cihazları, internet, firewall konularında deneyimli. - İngilizce bilen. Teknik döküman takip edebilecek düzeyde ingilizce bilgisine sahip - Evinden çalışabilecek. (Home ofis) - Çözüm odaklı olan, analitik düşünme yeteneği gelişmiş, sorumluluk bilinci yüksek, - Yoğun iş temposuna ayak uydurabi...

Gelen Voip Çağrıları Cevaplayarak Ses mesajını dinleyip kayıt yapacak ve gelen mesaj karşısında cevap dinletecek. Eş zamanlı 500+ üzeri çağrıyı cevaplayabilmeli C# Dilin de winform uygulama olacak kaynak kod için teklif istiyorum

Merhaba Nelly A., profilinizi fark ettim ve size projemi teklif etmek isterim. İstediğiniz ayrıntıyı sohbet üzerinden tartışabiliriz.

Merhaba Nelly A., profilinizi fark ettim ve size projemi teklif etmek isterim. İstediğiniz ayrıntıyı sohbet üzerinden tartışabiliriz.

Hi thepasserby82 Merhaba Ankara'da bir şirketiz.,Bir VoIP-SIP programı geliştirmek istiyoruz. Farklı olarak speech to text ile entegre olacak. Bunun örnek kodları github ta mevcut. Bunu imkan veren en iyi voip open source çözümü olarak asteriks bulduk. Türkiye de biri ile çalışmak bizim için daha avantalı olacaktır. İletişim bilgilerini paylaşırsan daha ayrıntılı bir şekilde konuşabiliriz. İyi çalışmalar. NOT: Bütçe miktarı ve süresi örnek olarak dolduruldu. Mehmet ARSLAN Computer Engineer| Manager | Co-Founder mhmttaa@

Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilmesi.

Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilmesi.

ekte görseli bulunan paranın modellenip animasyonun yapılmasını istiyoruz. Paranın iki tarafı aynı olacak ve kendi etrafında dönecek . Animasyon loop olup video H.264 codec alpha mov olarak teslimini istiyoruz.

...Border Controller (ASBCE) - specifically, to set up and integrate it with SIP Trunk Provider lines and Avaya IP Office Server Edition. AVAYA IP Office servers are already configured and calls working internally. - The tasks involved are both the installation of the ASBCE and the setup and integration with the SIP Trunk Provider. This requires a comprehensive understanding of both Avaya systems and SIP technology. - The primary goal for this configuration is to achieve a seamless integration between the ASBCE, the SIP Trunk Provider, and our Avaya servers. The focus is on ensuring robust communication and operation between these systems. - Basic training on how to restrict external call access to everyone, give dedicated SIP line to one user etc. Th...

I'm looking for a professional...me set up Audiocodes with Microsoft Azure Direct Route to integrate with a PBX system. Key Requirements: - Extensive experience with configuring Audiocodes for call routing - Proven track record in setting up SIP trunking - In-depth understanding of Microsoft Azure, particularly with Direct Route - Ability to seamlessly integrate Audiocodes with a PBX system The primary goal of this project is to ensure that the Audiocodes system works efficiently and effectively with the Microsoft Azure Direct Route, specifically for integration with a PBX system. You should have a solid grasp of SIP trunking and the ability to deliver a setup that guarantees optimal call routing within the integrated system. Note : i need with documents how to do so...

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

I'm seeking a professional who can configure my Kamailio SIP proxy and RTPengine. The main purpose of this project is to set up call routing and load balancing. Key Responsibilities: - Configure Kamailio SIP proxy for call routing - Implement load balancing features - Ensure compatibility with SIP over UDP Ideal Skills: - Proficient with Kamailio SIP proxy and RTPengine - Strong understanding of call routing and load balancing - Familiarity with SIP over UDP I do not require integration with any specific external systems or services. The primary goal is to optimize my setup for efficient call routing and load balancing. Please share your relevant experience when applying. It should read from a text file, xml or database the list of servers it nee...

I would like a Network Admin to configure the router of an internet network to allow the transmission of video signals from an RTP/RTSP-RTMP encoder, installed on a PC with a router equipped with a dynamic IP address in OFFICE A, to the recipient OFFICE B , which is connected to a server with a router with a static IP address. Reception/decoding in OFFICE B must take place via an RTP/RTSP-RTMP decoder or via VLC. We are also evaluating the possibility of implementing a CDN or a VPN to further optimize signal transmission.

...integrate a SIP telephony client within the Milestone video recording software as a plugin using the Milestone SDK. This SIP client will allow the configuration of connection data with the PBX and enable the functionality to automatically answer calls so that they are recorded by Milestone. Project Scope Development of the SIP Client: Implement a SIP client that can be integrated as a plugin into the Milestone software using the provided SDK. Ensure the compatibility of the SIP client with current and future versions of the Milestone software. SIP Client Configuration: Add a user interface in Milestone to configure the SIP client. Configuration parameters should include: SIP server address. Connection port. Username. Password. A...

I am in need of a highly capable and experienced iOS developer with demonstrated experience with the Linphone app. Key responsibilities: - You will be responsible for implementing major modifications to the existing Linphone app. - The focus will be to enhance communication services, so familiarity with VoIP and SIP protocols is necessary. - As I have a set deadline, timely completion is critical. Ideal candidate skills: - Proficiency in iOS development - In-depth knowledge of the Linphone app - Experience in enhancing communication services on VoIP apps - Demonstrable ability to deliver on tight deadlines Your experience in modifying the Linphone app, enhancing communication services and delivering within tight deadlines will be a great asset to this project.

I need the most simple Softphone/SIP app with my logo to use or another reseller voip/phone number provider so we can make calls in/out. I need this for Android, IOS, and PC/Mac. Simple app to call in/out. I need a custom softphone app (cross-platform) developed urgently. The app should be a simple dialer for both incoming and outgoing calls, and should be connected to servers. It's important that the app displays our company logo prominently. Key Requirements: - Development of a cross-platform softphone app - Integration with servers - Basic dialer functionality for both incoming and outgoing calls - Display of our company logo Ideal Skills: - Experience in softphone app development - Proficiency in cross-platform development - Ability to integrate with

I'm in need of a SIP and Asterisk expert to help troubleshoot issues with a SIP interconnect between two companies, focusing purely on voice communication. Key responsibilities include: - Identifying issues within the existing setup - Repairing and configuring the SIP and Asterisk system Ideal skills and experience include: - Proven experience with SIP and Asterisk systems - Strong troubleshooting abilities - Understanding of VoIP technologies - Excellent communication skills

...will give you: 1. Virtual Numbers 2. SIP Trunk server I am in need of an expert who can develop a SIP Trunk server with the capability of enabling live voice stream access. Key Requirements: - SIP Trunk Server: The primary purpose of this project is the development of a SIP Trunk server. It is essential that the chosen freelancer has experience in developing such systems. - Live Voice Stream Access: The server must offer functionality to enable live voice stream access. This feature must be robust, reliable, and seamlessly integrated into the server. - Access Control: I need the live voice streams to be secured with user-based permissions. This ensures that only authorized users can access the streams. Integration: - The primary platform that the SIP...

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

We have a voip app, voice calling with sip and fcm setup for the push notifications, we like to have a call alert like whatsapp calling alert that can handle both foreground and background states in Android and iOS. Also mention if somebody already worked on this type of feature, bid and connect to show a demo. Feature Required: Call Notification Incoming Screen Alert on App background and foreground State both killed or behind other apps. When you accept the call redirect to the ongoing call screen and call get connected with sip.

...from ongoing voice communication. Upon recognition of specified words, the code will need to send DTMF commands via a SIM7600 or SIP/VOIP system. In order to accomplish this project's requirements: - Proficiency in Python 3.5 is a must, as the script will need to be written fully in this version of Python. - Familiarity with serial communication is key, as this will be the method through which the script interfaces with the SIM7600 or SIP/VOIP system. - A thorough understanding of Google Cloud Speech-to-Text API is essential given its integral role in detecting specified words from speech in real-time. - Experience with telecommunication platforms, particularly SIP/VOIP systems and SIM7600 modules, will be highly beneficial for successful completion of this ...

I'm currently in need of a tech-savvy individual to set up the VLAN on my managed TP-Link switch for configuring the SIP line from Tata. Key responsibilities include: - Setting up the VLAN on the switch . - The task will be performed virtually; access will be given using Anydesk. Ideal freelancer should: - Have extensive knowledge of managing TP-Link switch for VLAN configuration - Be proficient in understanding networking concepts related to VLANs

I'm looking for a skilled engineer to create a CompactFlash CFast Digital Audio Player prototype. Qualifications: -must have Distortion Analyzer, frequency sweeps -be able to build a prototype board totally functional -Simulate Analog / Digital signals Regarding hardware Cflash Audio Player using AAC codec Full prototype hardware to be shipped to US Audio player will use CFast card via SATA (not USB) and play audio files using AAC codec DDDAC1794 circuit Dual Dedicated PCM clocks via TCXO 44.1 khz and 48 khz auto select MUSES IC attenuator Steel enclosure (pick something from China but big enough to support batteries) RCA output Remote

Urgent Loocking one program for change voice in live call / Need this program working on windows for softphone (zoiper ,xlite , eyes beam ) Its will be for call center ! Sip Agent need to change his her voice immediatly and voice must be natural . Girl to man /man to girl Im pay direct 30 usd who find me this program . And propose direct on test .

...efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice transit facility with incoming and outgoing SIP trunks. All incoming caller ID’s will be replaced by the DIDs provided for each MNO trunk. Connections 1) Multiple incoming customer Accounts from Soft switches like VOS3000 2) Multiple Outgoing SIP trunks provided by local MNO’s DID assignment/Man...

Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe

We have a VoIP app in flutter, voice calling with sip and FCM setup for the push notifications, we like to have a call alert like WhatsApp calling alert that can handle both foreground and background states in Android and iOS. If somebody already worked on this type of feature, bid and connect to show a demo.

I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..

I have Patton Smartnode which gets SIP line in (works) and needs to output to its 4 BRI ports to connect to classic PBX (Avaya IP Office 500). SIP line works BRI Configuration to connect Patton and PBX not, this is where i need help for.

We have a VoIP app in flutter, voice calling with sip and fcm setup for the push notifications, we like to have a call alert like whatsapp calling alert that can handle both foreground and background states in Android and iOS. If somebody already worked on this type of feature, bid and connect to show a demo.

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 providers: Your knowledge of di...

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.

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...

...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 as Call Recording, Ca...

...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 SIP t...

...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 SIP t...

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.

...looking for a skilled engineer to create a CompactFlash CFast Digital Audio Player prototype. Specific features and requirements include: Qualifications: -must have Distortion Analyzer, frequency sweeps -be able to build a prototype board totally functional -Simulate Analog / Digital signals Regarding hardware Cflash Audio Player using AAC codec Full prototype hardware to be shipped to US Audio player will use CompactFlash and play audio files using AAC codec DAC is PCM1794 or WM8742 in non-upsampling mode Dual PCM clocks 44.1 khz and 48 khz auto select MUSES IC attenuator Steel enclosure (pick something from China but big enough to support batteries) RCA output Remote This project requires deep familiarity with audio technology, battery operations, and prototyping. ...

skilled ffmpeg expert to work on a live transcoding project. - CMS should be built in laravel. it should use nginx for hls streaming with user pass and ip restriction protection. it should have all encode decode option...also option to select from wich gpu wich channels are going to transcode. it should have realtime datalogics like incoming outgoing network metrics realtime each gpu utilization of encoder/decoder graphic based utilization of ram and cpu. page with show all live streams/offline streams with options restart stop delete edit buttons streams should show with realtime datas like uptime speed codec bitrate connected users on stream count. user management for accessing live streams possible to groupbased management for adding streams in groups and acces grant lik...

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.

- Experience with multiple multimedia libraries and frameworks: FFmpeg, GStreamer, libvpx, x264, x265, etc. - Knowledge of multiple streaming protocols (RTMP, RTSP, RTP, HLS, WebRTC, DASH, etc.) and codecs (AAC, Opus, H264, H265, VP8, VP9, AV1, etc.). - Experience with at least one commonly used media server software: RED5, Wowza, Kurento, etc. - Highly proficient in database design, with both relational and NoSQL (Scylla, Redis) databases. - Comfortable working with AWS, Linux, Docker, Kubernetes, continuous deployment workflow, multiple programming languages tech stack.

...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...