Asterisk sip softphone工作

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    2,000 asterisk sip softphone 份搜到的工作,货币单位为 USD

    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    voip语音网关模块 已经结束 left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.

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    VoIP Video IM等即时通讯功能。 支持Android,IPhone,Window Phone等客户端应用。 熟识SIP Webrtc XMPP。 提供Demo。

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    主要指导elastix asterisk 的安装配置工作,熟悉Asterisk的模块(包括体系结构,配置文件,Log日志,History),能重新打包ISO文件

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    ...3、要求操作简单,点击“通话”功能按钮即可实现通讯; 4、由于即时通讯双方可能位置很远,因此必须经过互联网传输,需要中转服务器进行语音媒体的中转。中转服务器可以采用成熟的第三方软件或者协议,但是在项目完成后可以交付给我们; 5、发送方网络的最低可能是ADSL 2M。但是延时不能超过1S; 6、语音质量要高,不能有杂音、回音等问题,需要可以用于商业行为的那种; 7、需要有可供调用的组件和调用方法,我公司可通过调用方法进行上层功能的开发; 8、需要提供所有源代码和演示版本(演示版本的运行平台包括android、IOS),以及可交付的中转服务器; 9、并不需要完整的语音通讯软件,只需要能够实现语音传送、播放的功能模块即可。 10、演示内容:A用户点击“通话”按钮,中转服务器可通过设置好的IP地址,找到B用户,并发送通话请求。B用户接受请求后,双方可以进行通话。 11、你可以采用第三方组件,例如SIP,但是需要告诉我们是哪家的,效果如何。因为考虑到我国网络问题,所以建议是自己搭建服务器。 12、我们已经开发的软件可以在安卓、苹果商店找到,叫IEMAKER,通过安卓盒子,可以实现直播,但是目前还不能实现声音直播,因此,需要开发这个模块,用于该软件。

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    I'm in need of an Asterisk developer experienced with Asterisk 13. The primary task is to configure the system as a VoIP server for internal communication. Key requirements: - Understand Asterisk 13 well - Strong experience in setting up and configuring VoIP servers The ideal candidate should have: - Proven experience in customizing Asterisk for similar purposes - Good understanding of VoIP technologies and protocols - Ability to work independently and deliver high-quality work on time Please note that the project involves setting up the system, configuring call flows and ensuring it's ready for internal use.

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    Asterisk WebRTC Integration 5 天 left
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    I require an experienced and proficient developer who is familiar with both WebRTC and Asterisk. The objective is to expand the functionality of my current Asterisk setup by integrating it with WebRTC. The specific features to be integrated are: -Voice Calling -Video Conferencing This project seeks to enhance the communication capabilities of my Asterisk setup, thus the ideal candidate would have considerable experience managing such integrations in the past. Sound knowledge of VoIP, SIP, and related technologies will be greatly appreciated. An understanding of the Asterisk framework and its API will be critical.

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    I am in need of an expert who can effectively integrate WebRTC and Asterisk to achieve real-time audio and video communication. This functionality is intended primarily for internal team members. Therefore, it should ensure: - Seamless and effective call routing and handling - Successful integration with our existing telephony systems - Provision of standard-definition quality for both audio and video communication The ideal candidate for this project has substantial experience with WebRTC and Asterisk, and has successfully executed similar integration projects in the past. This professional should also be knowledgeable about telephony systems and their integration for real-time communication capabilities.

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    ...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...

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    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...

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

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

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

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

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    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...

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    ...each agent individually: - Other agents cannot see each other's campaigns. 4. Ability for the main agent to assign one campaign to all of their agents. 5. Exporting statistics by agents: - Export a text (or any other) file with answered/unanswered numbers specifically for that agent. - Export statistics for a specific campaign or overall for the day. 6. Integration with Magnus Billing based on Asterisk 13: - Login credentials for the auto-dialing module correspond to Magnus Billing users. 7. It will be web-based. 8. Easy connection to another Magnus Billing server and transferring the module to another server. 9. Ability to manage AutoDialer users Create a new user, delete an old one, disable a user Please provide a brief overview of your relevant experience and how you...

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    I'm seeking assistance with an urgent issue that's emerged with my softphone engine. - **Operating System**: My system is running on Windows. - **Programming Language**: Unfortunately, the programming language of my softphone engine has been omitted. I believe it may be C#, Java, or C++. It's imperative that the freelancer is comfortable with these languages to handle the issue. - **Timeline**: I'm in need of immediate assistance to solve this problem. Ideal candidates should be able to troubleshoot softphone issues and have experience with Windows software development. Experience with C#, Java, and C++ is a plus. The ability to act quickly and efficiently is essential.

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    Objective Develop and 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. Us...

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    ...As we fire up our grills and make vegetable salads for our summer barbecues, we know that we can safely consume cured meats and veggies as long as we maintain moderation. The preservatives, both natural and synthetic, help keep our foods safe from harmful pathogens so we can enjoy our meals without worry. The food label will state that there are “no nitrates or nitrites added,” but an asterisk will often lead to a fine-print addendum with the clarification, “except those naturally occurring in celery juice powder,” sea salt or a vegetable juice. As a result some “natural” or “organic” roast beef and turkey breast, or other products cured with sea salt, evaporated cane juice, potato starch, or natural flavorings or seasoning...

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    IVR Menu Capture Tool 已经结束 left

    I am in need of a simple, yet powerful tool that will enable me to capture and analyze the IVR menu options of my clients' toll free numbers. This tool will serve the ...Currently one of my team member is dialing the client's phone number, listening to the IVR options and manually capturing the details as follows *Level 1* Press 1 for English Press 2 for Tamil Press 3 for Hindi *Level 2* Press 1 if you are an existing client Press 2 if you are calling us for the first time *Level 3* Press 1 for sales Press 2 for Accounts Press 3 for service We already have a asterisk dialler in place and have the capability to dial phone numbers and also record the conversation but we don't have the ability to listen to the IVR options, dial each option to drill down to each ...

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    VoIP Server 已经结束 left

    I will need a voip server for my call center, I need it to be created in magnusBilling in a VPSand for my clients to be able to change the caller ID directly from a softphone using a code, I also need my clients to be able to play hold music, Will like to find someone want to work long term for the maintenance of the server

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    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.

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    I need a skilled Kamailio specialist to set up a proxy server that's capable of sending push notifications to Android and iOS softphone apps (We already have the apps). Key Requirements: - Cross-Platform Compatibility: The server needs to support both Android and iOS. - High Volume: The proxy will need to handle a substantial amount of push notifications, so performance and reliability are paramount. Ideal Skills & Experience: - Expertise in Kamailio: Experience working with Kamailio in a similar capacity is essential. - Push Notification Implementation: A strong understanding of push notifications and how to implement them across different platforms. Please provide details of your experience with Kamailio and any other relevant technologies in your proposal.

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    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 i...

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    PBX / VOIP / SBC engineer 已经结束 left

    I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.

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    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

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    ...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...

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    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.

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    I'm in need of a skilled professional who can set up a virtual audio mixing station for my radio. This will be specifically for Windows, so deep knowledge of this platform is essential. We use SoftPhone Xlite 3.0 to call clients We need someone to show us how can we play audio file and that playing will be set as Microphone on xlite but after i play voice, i will need to be able to speak from microphone without need to change settings how can we achievve that? Key Requirements: - **Experience**: I need someone with a good amount of experience in setting up virtual audio mixers for radio stations. - **Windows Proficiency**: As this setup will be on Windows, you should be well-versed with this operating system. - **Intermediate Level**: I need an intermediate level of comp...

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    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.

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    I'm in need of a knowledgeable and experienced Asterisk expert for the implementation in our company We need an Asterisk specialist to help us set up a carousel of different numbers and provider accounts. The essence of the task, when calling, automatically should be substituted random number from our list. Details and scheme will be discussed individually

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    ...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 ...

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    I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.

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    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

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    I'm seeking a proficient Asterisk developer who can build a custom PBX system with Issabela or something comparable. The PBX system requirements are as follows: - Less than 10 extensions. - Features such as call recording, auto-attendant, and call routing. - Compatible with VoIP phone lines. Ideal candidate should possess a deep expertise in Asterisk and a good understanding of VoIP technology. Knowledge of Issabela or similar software is an added advantage. Let's connect if you can guarantee a seamless and efficient communication system. For more information, sent the requerimients - The calls received would be answered by several receptionists who would log in/unlog in with their landline to receive the call. If there are more calls than receptionists, a message w...

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    As a sysadmin developer, I'm in need of an asterisk specialist to build a Docker Compose script or a bash script for an interactive vocal server. This project is multifaceted, carrying out outbound calls and saving responses in a database. Key responsibilities are: * Creation of an Interactive voice response (IVR) system. * Outbound calling function connected with my API for automated scheduling of phone calls. * MySQL database integration to securely store the recorded responses. For this assignment, it would be ideal if you have proficiency in using Asterisk, Docker Compose, API integration along with comprehensive database management skills It would be a cherry on top if you have prior experience constructing interactive vocal servers. Let's connect to disc...

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    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 .

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    ...professional who can install Asterisk PBX to facilitate a call routing system. Key Requirements: - Asterisk PBX will function primarily as a call router, modifying incoming caller ID's to the outgoing trunk DIDs. - The system should handle both incoming and outgoing calls 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 fa...

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    Asterisk PBX 已经结束 left

    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

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    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.

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    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 ..

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    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.

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    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.

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    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...

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    ...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...

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    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.

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    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.

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    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...

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    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.

    $19 / hr (Avg Bid)
    $19 / hr 平均报价
    6 个竞标