Sip softphone solarispekerjaan

We used Asterisk as our phone system And we used Flutter's Sip-UA as a client The communication platform is WebRTC The problem we have is that when the internet suffers a few packet losses during a call, the client leaves the channel and then it is completely silent until the call is disconnected. We simulate the same scenario with a softphone, after the internet is disconnected and reconnected, the call continues and there is no problem. Are there any friends who can guide me?

I am in search of a proficient developer to apply branding elements to a Linphone Android & iOS app. The elements include: - Incorporation of an existing logo into the app - Placement of a brand logo on the app's homepage Required skills and experience: - Proficiency in mobile application development, specifically Android and iOS - Expertise in integrating visual branding elements into mobile applications - Experience working with Linphone apps would be highly beneficial The ideal candidate should have the ability to place the logo on the top right corner of the app's homepage. Your application will be favored if you provide samples of your previous work with Linphone apps or similar. Looking forward to your bids!

I'm looking to add SIP telephony functionality to my existing Windows Forms project. Here's what I need: - A feature to make and receive phone calls through the application. This is essential to the effectiveness of the tool as it is primarily a communication enabler. - I only need to support a single line. Multi-line telephony is not necessary for the scope of this project. - An added functionality to provide real-time call statistics is required. It is crucial to get insights into the ongoing calls which can later be analyzed for quality management and improvements. Ideal candidates for this job are those with a strong background in .NET and Windows Forms development, as well as a clear understanding of VoIP protocols, particularly SIP. Knowledge in implementi...

I'm looking for an expert to set up advanced configuration for the Grandstream UCM6302A SIP. This involves deep knowledge of SIP protocol, VoIP technology, and complex call routing. The ideal candidate would also possess strong Lua programming skills specifically in manipulating incoming calls using an external API on the Grandstream device. Key Responsibilities: - Full, advanced configuration of the Grandstream UCM6302A SIP. - Implement script in Lua for incoming call handling via external API. Ideal Skills: - Advanced SIP configuration - Expertise in VoIP technology - Proficient in Lua programming - Previous work with APIs. Previous experience with Grandstream and Lua is not a necessity, but a strong understanding of both is required for project ...

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I am seeking a skilled animator to create an engaging video presentation that showcases the company profile and history AND the product features and effectiveness . The video should cover all key features including: **Narration not required! Video length: ~150 sec Description: The company sells SIP panels with high energy efficiency. Clients are stakeholders in the real estate development/construction companies. The panels (simple square shapes, like the ones in video below) should be animated to highlight the assembly process (simplified) and the two components they are made of. Deadline is 14th March 2024 Video Example (quality should be at least as good, preferably better):

I am setting up asterisk with Amazon Chime for sip trunking, but am very new to this and am having difficulty where making a test call doesn't work. If you think you can help, please bid the amount you would charge, and use the word "alarm clock" somewhere in your bid so I know you read the description and the bid isn't automated. If things go well, I will likely need more help in the near future setting up other details of the Asterisk config. For now though I am just trying to get a test call to work.

Hi, I'm looking for a working code or freelance who has integrated the Linphone SDK from native to Flutter, Android and especially in IOS. Currently we already have sip server ready to connect with SDK, and need flutter working code to run in iOS with pushKit and callKit in background and foreground and also in Android. Current Sip Server have Mizutech VoIP install and done setup. Module Code in Flutter: 1. Voice Call 2. Video Call 3. DTMF 4. Chat

ONLY APPLY IF YOU HAVE EXPERIENCE IN VoIP and SIP. I am in need of a skilled freelancer with intermediate experience in Asterisk to set up a VoIP and SIP trunk on AWS. The successful candidate should have solid background with AWS infrastructure and Asterisk application. Key Deliverables: - Set up VoIP on AWS using Asterisk - Implement a SIP trunk functionality Requirements: - Must have experience in setting up and managing Amazon Web Services (AWS). - Asterisk application experience is essential. - Previous work on similar projects will be advantageous. Please make sure to include your experience with Asterisk and AWS in the application. I look forward to receiving your proposals and working with you on this exciting project.

I currently have a VoIP network which needs extending and enhancing. I'm seeking professional assistance in order to add a SIP trunk on my FusionPBX and implement automatic call distribution (ACD). Your tasks will include: - Adding SIPstation as my SIP trunk provider on FusionPBX. - Implementing automatic call distribution for enhanced call management. The ideal freelancer should have: - Extensive experience with FusionPBX - Profound understanding of SIP trunking - Familiarity with SIPstation and its integration to FusionPBX - Proven ability to setup Automatic Call Distribution. I look forward to receiving your bids and working together to advance my VoIP setup.

...currently in need of a talented artist who can help me bring to life a Women's Day themed sip 'n paint event. I am specifically looking for art that combines: - Feminine Symbols: Depictions of classic feminine symbols with a fresh perspective are requested. - Empowering quotes: Integrate quotes that celebrate and uplift women into the artwork. They should be beautifully woven into the design and not simply an add-on. - Women's Rights Icons: Depictions of iconic figures that symbolize women's rights also need to be a part of the theme. The preferred color scheme is blue and gold. The artist should work with these shades primarily but isn't restricted to just these colors. Skills and experience: - Experience with sip 'n paint styled events or...

I’m seeking a proficient developer with a deep understanding of video streaming, specifically in H.264 codec streaming and VoIP SIP communication. The objective of this project is to develop a system based on pjsua that can read an RTSP stream of an IPC (Internet Protocol Camera) and transmit it to the other end of a SIP phone for display. Key Requirements: * Expertise in pjsua programming * Profound understanding of RTSP, H.264 codec, and VoIP * Ability to integrate IPC for video streaming * Design the system to work seamlessly with SIP phones * Ensure high-quality video streaming without any lag The developer I wish to hire should have a strong command of the software and technologies required for the project and should provide a demonstration of their past ...

I'm in need of an expert who is well versed with Grandstream devices for a pressing configuration task. This requires someone who can professionally handle SIP trunking configurations and ensures seamless routing. The objective here is simple — I'm seeking making my device scalable for future expansion. Ideal skills and experience required: - Comprehensive understanding of Grandstream devices - Proven experience with SIP trunking configurations - Proficient in delivering flawless routing solutions Preference will be given to individuals who have successfully completed similar tasks. Your assistance will be greatly appreciated!

... Operating the server is not part of your scope. • I am the only end-user. Therefore, the set-up should be tailored around my requirements and preferences. • As for functionality, I aim to utilize SIP trunking, Voice Over IP (VoIP) services, and Multi-Tenant PBX services with FreeSWITCH and FusionPBX. However, I'm interested in leveraging the complete capabilities these tools offer, so your input on additional beneficial features is welcome. To best fit this project, you should have demonstrable expertise with FreeSWITCH, FusionPBX, and Debian server environments. Familiarity with VoIP, SIP, and PBX services is also paramount. Please base your bid on a complete project execution, and if there are any specific details you need to know beforehand, don...

Hi there! Please if you could design us many options for our Sip and Shop Art Event! We have worked on a draft and that is attached below as inpiration! We have also attached other inspiration! Flyer to include what is in the first inspiration. All of the copy/text to be included is in the Sip and Shop Art Night Flyer PDF PLEASE SEE SIP AND SHOP ART NIGHT FLYER PDF FOR WHAT TO PUT ON FLYER:)

please teach me how to install, configure kamailio, trouble shoot and check log. I have a need for SIP Proxy

I'm seeking an experienced network engineer to work on my Cisco 2951 STC telephone line VoIP configuration. Key Responsibilities: - You will primarily be responsible for setting up VoIP using the SIP protocol. - Integration of specific SIP trunking features is also needed, in line with our operational requirements. Skills and Experience: - Prior experience in configuring Cisco 2951 STC or similar products is a must. - Extensive knowledge and experience with VoIP technology and SIP protocol. - Familiarity with SIP trunking features would be considered a great advantage. Candidates who can deliver the project promptly and professionally will be highly preferred. This project offers the opportunity to showcase your abilities in a complex, highly technical...

I'm seeking an experienced network engineer to work on my Cisco 2951 STC telephone line VoIP configuration. Key Responsibilities: - You will primarily be responsible for setting up VoIP using the SIP protocol. - Integration of specific SIP trunking features is also needed, in line with our operational requirements. Skills and Experience: - Prior experience in configuring Cisco 2951 STC or similar products is a must. - Extensive knowledge and experience with VoIP technology and SIP protocol. - Familiarity with SIP trunking features would be considered a great advantage. Candidates who can deliver the project promptly and professionally will be highly preferred. This project offers the opportunity to showcase your abilities in a complex, highly technical...

I'm seeking an experienced network engineer to work on my Cisco 2951 STC telephone line VoIP configuration. Key Responsibilities: - You will primarily be responsible for setting up VoIP using the SIP protocol. - Integration of specific SIP trunking features is also needed, in line with our operational requirements. Skills and Experience: - Prior experience in configuring Cisco 2951 STC or similar products is a must. - Extensive knowledge and experience with VoIP technology and SIP protocol. - Familiarity with SIP trunking features would be considered a great advantage. Candidates who can deliver the project promptly and professionally will be highly preferred. This project offers the opportunity to showcase your abilities in a complex, highly technica...

I'm ...specifically someone with experience in Asterisk and SIP integrations. Key Project Aspects: - Asterisk Setup: Your task will be to set up Asterisk with multiple features including monitor calls + transfer + hang up calls, an IVR system, and call recording. I require a high level of accuracy and efficiency in the configuration of these features. - SIP Service Connection: Additionally, I need to connect Asterisk to a SIP service. We have an AI that speaks and give response to your recorded message. So the idea is to connect this AI to the Asterisk that is connected to the SIP so it can speak with people on the phone. Ideal Skills: - Experience setting up and configuring Asterisk, IVR systems, and call recording - Knowledgeable about SIP s...

...inspector for a new construction project in Evergreen Colorado. This inspection requires an onsite visual inspection. To do so you must be in the proximity of Denver Colorado. The inspection will focus on both the SIP panels and steel welds. Here are a few key points about the project: - The key requirement is an unbiased, independent, and professional evaluation of the structural quality and compliance of both the SIP panels and steel welds with local building regulations. - Experience in conducting thorough inspections according to local regulations is essential. - Knowledge of SIP panels and steel welds is mandatory for a successful completion of the task. Your expertise will aid in satisfying the Jeffco building department's requirements for our projec...

We are in the early stages of creating an AI-powered call center service tailored for businesses in Egypt, such as restaurants and car dealerships. Our goal is to leverage GPT-4 integrated with AI text-to-speech and speech-to-text models to provide seamless, efficient, and cost-effective customer service solutions. We are seeking an experienced ...Frameworks (React, Angular, Vue.js) Node.js, Django AI and Machine Learning Libraries (TensorFlow, PyTorch, Hugging Face's Transformers) Cloud Platforms (AWS, Google Cloud, Azure) API Development Docker, Kubernetes CI/CD Tools (Jenkins, GitLab CI/CD, GitHub Actions) Web and Data Security Practices Problem-Solving Skills Communication Skills Project Management Skills Telephony Integration (VoIP, SIP) Speech Processing (Speech-to-Text, ...

Hi I have a working ipsec tunnel (one host as a client and one host as a server) and I need to make sip calls from asterisk (as a client) to freeswitch (as a server) thru this tunnel. It works outside the tunnel but when I try to use the ipsec tunnel IP I can't hear audio and get call hangup after few seconds. I can provide ssh to both client and server and it's easy to 'switch' between using and not using the tunnel. Please write down the changes that you make. Also please only bid if you have experience with freeswitch. Max 100 euros. Thank you.

I'm searching for a proficient SIP/Asterisk engineer who has ample experience in performing the following tasks: - Setting up and configuring an Asterisk server - Ensuring that the server is set up for high security and can support telemarketing or sales calls - Adding a specific function of calling and spoofing numbers, to be used exclusively for telemarketing and sales purposes Ideal skills and experience for the job: - Extensive experience with Asterisk and SIP protocols - Past-experience in setting up IVR systems - Understanding of server security measures - Familiarity with telemarketing systems Please provide a summary of your relevant work experience and your approach to this task in your bid.

I'm searching for a network security expert to assist in diagnosing a problem with my SIP ALG settings on my Cisco ASA5506 firewall. It's crucial because I am experiencing an issue wherein the phone rings for one second and then cuts off. Your tasks will include: - Examining the SIP ALG status on my firewall. - Troubleshooting any issues found and offering solutions to rectify them. - Implementing the solution with my approval. Relevant Skills and Experience: - Proven experience with Cisco ASA5506 firewalls. - Deep understanding of network security. - Skills in VoIP technologies and SIP ALG. - Familiarity with various VoIP service providers/software is a plus. Please note, we are not currently using Skype for Business or Ring Central, but a different prov...

I resolved issues with developed pbx based asterisk with dial plan and fax over sip

I will install Freepbx on a local server and in Digital Ocean Cloud. I will configure it for 10 users and 3 trunks, while also ensuring that configurations can be synced between them. Finally, I will secure it against external threats and SIP flood attacks.

I need an imaginative designer to create a logo for my Italian Bistro, SipsAndSpices, where flavor meets elegance. Our bistro offers a curated selection of fine wines, craft cocktails, and globally inspired dishes. With a warm ambiance and attentive service, we invite you to savor the exquisite blend of flavors and sip on carefully chosen libations. I am open to all styles, be it minimalistic, vintage, modern, or a unique blend of these. Color is also your playground; creative use of blue, green, yellow, or any other shades are welcome. Just bear in mind that the logo should reflect an Italian restaurant vibe. Skills and Experience: - Proficiency in graphic design software. - Knowledge of effective color schemes in logo design. - Ability to design within different styles. ...

I'm seeking an expert Android developer to create a VoIP app with GSM integration. This pro...developer to create a VoIP app with GSM integration. This project includes, but is not limited to: - Developing an app for calling and text messaging through a SIP server, inclusive of delivery reports. - Implementing features such as call recording and instant messaging. - Ensuring compatibility with older Android versions. Since I don't have a preferred SIP server, your input and recommendations would be highly valued. Experience with SIP servers and a strong understanding of both VoIP principles and Android development are necessary for this project. Your role will be pivotal in guiding the selection and integration of the SIP server. Please consider the ...

Salut, Pe lan 1 avem reteaua locala calculatoare si telefoane voip si internet Pe lan 2 avem acces la serviciul digi sip care la care sa conectat dar nu mai vedem telefoanele din retea initial vedeam telefoanele ba chiar am gacut si teste cu un modul GOIP care a functionat bine .

More details: What type of engineer do you need? Structural engineer Could you please clarify what specific applications of SIP panels are you interested in? Wall system, Roof system, Floor system What are the approximate dimensions of the cottage? Small (less than 500 sq ft) , 12 feet wide by 20 feet long ICF blocks for foundation and walls. Sip panels for floor, Gable ends for roof and roof engineering for attachments See attached file to read details

...essential configuration changes to my existing Issabel IP PBX setup. The project specifics include: 1. Changing the current SIP provider from Tata to Airtel. 2. Updating IP PBX configuration settings to seamlessly support the new Airtel SIP service. 3. Making necessary modifications to the SIP trunk settings to ensure optimal compatibility with Airtel's service. 4. Configuring additional SIP trunk registrations as required. The ideal candidate should possess proficiency in SIP trunk configurations, with a solid track record of experience in working with both Tata and Airtel SIP services. A clear understanding of Issabel IP PBX setups and the ability to configure SIP settings accordingly are crucial for the success of this project. ...

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I need a comprehensive telephony solution, utilizing both FreePBX and Asterisk servers to streamline incoming, outgoing, and call monitoring processes. Here's a glance at my requirements with the BSNL SIP Direct Truck : 1. FreePBX AND Asterisk Server Setup: I am looking for an experienced tech consultant who can not only advise but also assist in setting up both FreePBX and Asterisk servers. 2. Custom Bot Development: The bot should be responsible for calling, receiving, and monitoring calls directly from the software with features such as: - Automated call routing - Call recording - Call analytics - Call forwarding - Interactive Voice Response (IVR) - Real-time call monitoring Ideal skills would include a deep understanding of VOIP technologies, especiall...

Project Description: FreePBX and GOIP Configuration Looking for the initial setup of a FreePBX systems with GOIP GSM gateway for inbound trunks. Softphones will be used by internal clients to receive calls. Skills and Experience Required: - Experience with GOIP GSM gateways. - Proficiency in FreePBX configuration. - Softphone configuration. - Understanding of custom configuration setups, if necessary If you have the expertise in these areas, please bid on this project. Thank you.

...communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP trunk management. - Strong communication and problem-solving skills. - Preferred Qual...

instale astpp basado en astpp, el problema que presento es que luego de haberlo instalado en debian 10 no me permite realizar conexiones sip, cabe destacar que desactive el firewalld y fail2ban para ver si era el problema y aun no me permite realizar conexiones

Sales :- • Conduct market research to understand the financial needs and preferences of target customers. • Prepare sales materials, presentations, and proposals to engage with prospective clients. • Source new clients and get repeat business from existing clients. • Sell Investment products I.e. Mutual Funds SIP & Lumpsum, Bonds, FD’s.

I am looking for a Python expert, ideally with experience in VoIP, Call-center's Dialler softwares and SIP protocols. I need help in building a python application(just a script/server side python app) that will connect to VoIP/Dialler and get the audio stream for call and reply with an audio stream. I've a Dialler(MicroSip) setup on one of my servers, I have used the JS example and LinPhone application to make calls between 2 apps and it works fine. What i want to achieve is: Write a python program that should recieve, and make calls using computer's Michrophone and Speakers and should have the option to add anyother audio stream in reply to a call instead of getting the audio from system's michrophone. I've gone through plenty of libraries but as i am no...

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

...hardware - Implement functionality for sending commands to a remote M5Stack device over LTE - Enable the M5stamp LTE-M to connect to a web server and display online status - Integrate SIP/voice call capabilities from a PC to the M5Stack LTE-M remote device using the web application - Allow AT commands to be sent from a local PC application to the remote M5Stack Stamp-M Key Skills Needed: - Proficient in WebSockets for real-time communication - Experienced with Arduino coding specifically for M5Stack Stamp - Skilled in web application development, preferably with a focus on user-friendly interfaces - Familiarity with SIP protocol for voice communication over the internet - Ability to manage AT commands and their integration in software - Understanding of real-time data mon...

I'm seeking assistance in modifying an existing Python3 code to listen to incoming calls on a SIP server. At present, the code can only make calls. The ideal candidate for this task is experienced in Python3 and SIP server management. If your previous work includes similar projects, it will be beneficial in placing your bid. The necessary modifications include: - Adding a call recording feature - Implementing a call forwarding functionality - Integrating a call logging system While applying for the task, please consider that sharing details about your experience and past work is appreciated. Freelancers who can deliver the project quickly will be given preference. Nevertheless, I'm more interested in quality work, so feel free to negotiate a reasonable deadli...

...communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP trunk management. - Strong communication and problem-solving skills. - Preferred Qual...

...around fixing bugs in an Android app, specifically problems with its SIP functionality. The key issues I'm experiencing are: - The SIP registration automatically stops. This tends to happen when the app is runing . - Certain SIP functions related to incoming and outgoing calls and call quality issues are not working as expected. An ideal candidate would possess the following skills and experience: - Proficiency in Android app development - In-depth understanding and experience of VOIP and SIP communication protocol - Experience in troubleshooting and resolving SIP registration issues - Experience in debugging and problem-solving SIP related call issues. Your primary task will be to fix these bugs to ensure that SIP registratio...

I'm seeking a talented developer to create a reliable and secure softphone application for both Windows and Mac operating systems. Key Features Required: - Call Recording: Ability to record voice calls with clarity. - Video Calling: High-quality video call functionality. - Voice Encryption: Ensuring secure, encrypted voice communication. - Real-time Language Translation: Feature enabling audio SIP calls with instant language conversion. User Interface Design: - The interface should support customizable themes, allowing users to personalize their experience. Ideal Skills and Experience: - Proficient in cross-platform software development. - Experience with SIP protocol and audio/video encoding. - Knowledgeable in implementing encryption algorithms for voice secu...

We are using SIP account using Centos for Linphone ..TCP already configured however with 5g connection sometimes calling doesn't work. We need an expert who can setup/Enable TLS on it. Linphone sdks are installed on both iOS/Android and backend.

I'm seeking an experienced tech to setup both inbound and outbound trunks on my FusionPBX system. Please note this task involves setting up Inbound/Outbound Trunk using our Provider's IP and also you need to provide us step-by-step instructions to perform this in future. The ideal candidate should have: - Extensive experience with FusionPBX. - Solid understanding of SIP trunking, as I am currently using a SIP trunk provider. - A proven record of successfully setting up both inbound and outbound trunks. I am expecting a smooth configuration process with minimal downtime. Please ensure your bid reflects your understanding of these requirements. Maximum Budget - INR 3500 / $45 USD

Looking for a good software developer for an ongoing project to develop a robust softphone app that will connect to a SIP PBX and be able to make and receive calls. - Register to a sip server - Make and receive calls - Call History: The app must track and display complete call logs. - Contact Management: Users should be capable of managing their contacts within the app. - Voice Messages: In-built feature allowing users to send and receive voice messages is required. We look to start with one OS first, most likely iOS. Then if the project is successful we look to develop the same app for Android. If both are good we would like to include Windows. Ideal Bidders: Developers with experience in VoIP app creations, cross-platform development and understanding of VoIP ser...

un sistema de llamadas masivas con troncal SIP

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.