Pjsip softphone jobs
I am seeking assistance with outbound telesales in the fashion recruitment industry, with the core objective of generating high quality leads, and setting follow-u...will forward hourly. There is potential to make this an ongoing job for the right Freelancer that can achieve daily and weekly targets. Daily target is to deliver 20 high quality qualified leads (name, email address, direct contact number) Weekly defined bonuses for the delivery of aggressive targets also available. Working hours will be 5 hours each day. Calls will be made via internet softphone. You should have professional level headset. Salary is $7 per hour, paid daily in arrears on delivery and verification leads.. We will be conducting initial interviews within 24 hours. Successful candidates can start...
...developer who specializes in VoIP and PJSIP development. The successful candidate would 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 u...
I'm looking for an expert developer well-versed in Twilio and CRM systems to integrate Twilio's voice calling feature into a softphone for a desktop environment. Integrate with BREA and make it our default dialer for any phone as click to dial. It must be able to reeive and send a call. Key Responsibilities: - Integrate Twilio's voice calling feature with our existing CRM system in a manner that optimizes user experience and call quality - Ensure the softphone functions seamlessly on our desktop platform Ideal Skills and Experience: - Proven experience with Twilio's voice API - Past work on softphone setup and integration in a desktop environment - Strong understanding of CRM systems - Excellent troubleshooting and debugging skills Your exp...
...developer who specializes in VoIP and PJSIP development. The successful candidate would 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 u...
...experiencia configurando la libreria de Javascript SIPML5 y asterisk. Tenemos todo instalado y configurado. El softphone web se registra al asterisk, emite y recibe llamadas, pero cuando se atiende la llamada no transmite el audio. El servidor tiene instalado una VPN. Cuando el usuario se conecta a la VPN, entonces funciona el audio de la comunicación pero cuando no se conecta a la VPN entonces vuelve el problema del audio. Se requiere que el softphone funcione sin VPN, solo por internet. Version asterisk 18 OS: Ubuntu Server 14 +++++++++++ A person with experience configuring the SIPML5 and Asterisk Javascript library is required. We have everything installed and configured. The web softphone registers to Asterisk, sends and receives calls, but when the ca...
I'm seeking a skilled developer to create a versatile VOIP softphone app, compatible with both iOS and Android devices. Key Features and Functionalities: - Video calling - Conference calling - Ability to send and receive SMS & MMS messages - Auto provisioning function - In-app address book Further, ensuring seamless integration with APIs is a must, as well as a functionality syncing with an address book or contacts. The ideal freelancers for this project would have solid experience in iOS and Android app development, with special emphasis on communication apps and certificate pinning. OPENSIPS knowledge + Knowledge in API integration and VOIP technology is highly essential. Your portfolio showcasing related projects will highly increase your chances of being selected. A...
You can bid either by a price per 100 calls, or a price per 4 hour day. As a firm offering financial refunds in the Healthcare sector, we're specifically focused on securing energy refunds for our clients. We're in need of an experienced English-speaking phone appointment setter who can persuasively liaise and secure meetings with prospective clients in the healthcare industry. Y...effectively We need a dedicated professional who can not only understand the technical aspects of our services but also effectively communicate them to potential clients. If you have a stellar track record in appointment setting and sales, specifically in the healthcare industry, we want to hear from you. We have 5000 records to call , all based in the UK. Script will be supplied, as well as s...
I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50
As fast as possible, I need a professional to assist in setting up FreePBX. Key tasks essential for this project will include: - Installation and configuration - Call routing and extensions - Integration of FreePBX with a CRM system that is yet to be determined In addition, assistance is required for setting up VoIP softphone. Familiarity with popular CRM platforms like Salesforce, HubSpot, Zoho CRM is an asset, as one of them may be chosen for integration. Absolute proficiency in setup, call routing, extensions and integration with CRM systems is a must for this job. I am looking for a fast turnaround on this project, so previous experience achieving quick deployment is highly recommended. It is also necessary to be able to advise our mobile developer if he has questions
...accurately and keep a comprehensive database of contacts, which you will forward hourly. There is potential to make this an ongoing job for the right Freelancer that can achieve daily and weekly targets. Weekly defined bonuses for the delivery of aggressive targets also available. Working hours will be between 3 to 6 hours each day, to be agreed, based on timezone. Calls will be made via internet softphone. You should have professional level headset. Salary is $8 per hour, paid weekly. To apply, please send over your up to date CV/resume, and a voicenote or wav file detailing 1. your most relevant experience, 2. your best successes achieved in the last year, and 3. a summary of your three best qualities. We will be conducting initial interviews within 24 hours. Succes...
I'm in need of an experienced developer to design and build a softphone application. Key Project Requirements: - As development requirements have not been specified, it is assumed that the application should be compatible across all main operating systems like Windows, MacOS, and Linux. - Softphone's features weren’t specified, but usually, a robust softphone should have features like call recording, conference calling, and instant messaging. Ideal Skills and Experience: - Proven experience in creating effective softphone applications. - Strong knowledge of cross-platform development. - Familiarity with voice-over-IP (VoIP) technologies and protocols. Application Instructions: Even though not specified, it's generally best to include your pas...
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?
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.
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 security. - Famili...
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 service providers...
We are seeking a skilled freelancer with expertise in Asterisk to assist us in implementing outbound call functionality to a mobile number using PJSIP. The current channel originate command is not working. and we are open to alternative methods that successfully initiate calls. Earlier, when we used older version of asterisk 18.1, we can get calls but newer version doesnt have that command and it gives below error pi*CLI> originate sip/ application Playback hello-world No such command 'originate sip/ application Playback hello-world' (type 'core show help originate sip/' for other possible commands) We are using asterisk 20.5 and system is rpi with ubuntu 23 installed. Thank you!
...We prefer Twilio. But are not sure if Twilio is offering a Sip to Whatsapp Gateway service - You will build the script / library that enables asterisk to terminate calls to a whatsapp endpoint. - Asterisk is build from source. - A AWS Vps to work on can be provided Here is what the dial plan code could look like that needs the extra option for whatsapp calls same => n,Dial(PJSIP/+${outboundNumber}@${trunkprovider}&PJSIP/${whatsappNumber}@${whatsappTrunkPriver},30) Deliverables - Documentation on how to build asterisk from source with extra libraries etc - A proof of concept that it works. Ideal skills and experience for this job include: - Strong knowledge and experience with Asterisk call termination - Experience with integrating Asterisk with WhatsApp - Familia...
Looking for Experienced Freelancer to Customize Linphone SIP Softphone! About Us: We are a dynamic and forward-thinking company seeking a talented freelancer with expertise in customizing Linphone SIP softphone applications. As we strive to enhance our communication solutions, we are in need of a dedicated individual to make specific modifications to our Linphone app on both Android and iOS platforms. Project Scope: We require a skilled freelancer, not an agency, to handle the following tasks: Design Customization: Transform the existing Linphone SIP softphone design to align with our brand identity and aesthetic preferences. Implement a sleek and user-friendly interface that resonates with our vision. Push Notifications Fix: Address push notification issues on bo...
Complete Solution for GoIP Hardware (GSM Gateway + SIM Bank) Budget: More than $1000 Primary Purpose: Automation The main goal of a project is a creation of software platform to manage a number of distributed network of GoIP GSM gateways and SIM Banks (SMB). We should be able to make mass calls, SMS/USSD bulk sending & receiving. Asterisk PBX must be used as...SIM/channel enable/disable/restart - IMEI display/modify - signal level display - human behaviour simulation (to increase SIM cards lifespan) - bulk SMS/USSD sending & receiving - automated SIM cards check, kick dead (blocked) SIM cards if detected - live tracking of ASR and ACD for each SIM card and channel - live total / answered / failed calls per SIM/channel Asterisk part must be built on top of PJSIP and ARI (...
I am looking for a React-Native (Expo) developer to create a SIP SoftPhone App with advanced calling functionality. The app will NOT need to integrate with any existing systems or APIs. Ideal skills and experience for this job include: - Proficiency in React-Native development - Strong understanding of SIP protocol and VoIP - Experience with building calling applications - Ability to create a user-friendly and visually appealing design If you are confident in your skills and can deliver a high-quality app with advanced calling features, please apply. The app needs to be active in background, just like normal native phone app. And needs to be always avaialble to receive calls. Only thing we will provide are: SIP URI & username/password. Platforms to support: - Android...
CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHN...connection is made, then send messages between some clients and then terminate the app. This is just the basic summary. Please refer to the PDF attached for the full specification. I don't want to use WebRTC due to its high latency, but instead need to use raw UDP packets (SOCK_DGRAM). This is both portable and performant. There are various NAT traversal libraries out there, but I prefer using PJSIP because Android and iOS devices are supported. If you prefer to use another library, please consult me first. Thank you CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY ...
CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHN...connection is made, then send messages between some clients and then terminate the app. This is just the basic summary. Please refer to the PDF attached for the full specification. I don't want to use WebRTC due to its high latency, but instead need to use raw UDP packets (SOCK_DGRAM). This is both portable and performant. There are various NAT traversal libraries out there, but I prefer using PJSIP because Android and iOS devices are supported. If you prefer to use another library, please consult me first. Thank you CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY ...
CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY BACKEND WORK! DO NOT USE CHATGPT IN YOUR PROPOSALS. YOU WILL BE IGNORED. Hi I ...to a list of backup TURN servers. Once the connection is made, then send messages between some clients and then terminate the app. I'd rather not use WebRTC due to its high latency, but instead prefer using raw UDP packets if possible (such as sendto, recvfrom etc). Is this something that can be done in C++? I notice that CoTurn has a library you can use, but there are also other ones such as Pjsip. CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY...
I am looking for a freelancer who can help me configure Zoiper softphone on my VOS3000 switch. I am currently facing issues with calls and need assistance in resolving this. Additionally, I also require help with other settings on the VOS3000 switch - specifically confirming that customer account is set up properly. Ideal skills and experience for this job include: - Proficiency in configuring Zoiper softphone on VOS3000 switch - Troubleshooting and resolving issues related to softphone calls - Familiarity with VOS3000 switch and its various settings - Strong communication skills to effectively understand and address my requirements. I need this task completed in next 30 minutes. This is 3 minutes task and the maximum I will pay is $10.
I am looking for a developer who can help me make a connection between my client and a specific SIP server using the PJSIP module and React Native. Preferred Skills and Experience: - Proficiency in working with the PJSIP module and integrating it with React Native - Experience in setting up and configuring SIP servers - Strong understanding of networking protocols and security measures - Familiarity with voice call functionalities and implementing them in mobile applications Project Requirements: - Connect the client's application to a specific SIP server of their choice - Implement basic security measures for the connection - Develop the application to support voice calls functionality If you have the necessary skills and experience, please reach out to discuss fu...
I am looking for a freelancer who can help me with a project that involves using FREEPBX and asterisk ari to place outbound calls out of a sip trunk using pjsip. The purpose of the outbound calls is for customer service. I do not have any existing infrastructure to support this project, so it needs to be built from scratch. For project updates, I prefer communication through email. Skills and Experience Required: - Strong knowledge and experience with FREEPBX, asterisk ari, and pjsip - Previous experience with setting up outbound calls and sip trunks - Excellent problem-solving skills - Ability to work independently and meet project deadlines
We are seeking an experienced freelancer to facilitate the seamless integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the u...capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the utmost security for our call traffic through Encrypted Media. Project Requirements: -Integration of WebRTC functionality into the Isaaabel platform. -Configuration of JsSIP npm package to establish connections with our Asterisk calls to SIP and PJSIP extensions through the WebRTC interface. -Implementation of robust security measures, including encrypted media, to safeguard call communications.
I am looking for a freelancer to help me integrate a WebRTC based calling experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement s...Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement so it would be great if the freelancer can deliver the project quickly. Resume, we need to connect WebRTC extension with Issabel. I need that JsSIP npm package can connect with my asterisk server and make calls to SIP and PJSIP extensions. I need the documentation of the implementation for future re-install ...
Project Description: fusionpbx remote SIP account setup I have fresh installed fusionpbx at my LAN. It is hosted my odroid XU4 (similar to raspberry pi) at my home network. I cannot remote register to the fusionpbx out of LAN with softphone. I want one remote account set up. I am looking for a freelancer who can help me set up a remote SIP account on my fusionpbx system. Requirements: - Existing SIP provider: Yes - Devices: Both desktop phones and mobile devices will be using this SIP account - Level of experience required: Expert Skills and Experience: - Knowledge of fusionpbx system and its configuration - Experience in setting up remote SIP accounts - Familiarity with configuring SIP providers and integrating them with fusionpbx - Basic understanding of networking and VoIP...
I am looking for a skilled developer who can take the PortSIP SDK and create a softphone specifically for Windows operating system. The softphone should have the capability to record calls. Key requirements: - Development of a softphone using PortSIP SDK for the Windows operating system - Integration of call recording feature - Ensuring smooth and reliable performance Ideal skills and experience: - Proficiency in PortSIP SDK and Windows development - Knowledge of VoIP protocols and technologies - Experience in integrating call recording functionality - Strong problem-solving and debugging skills Timeline: The project needs to be completed within 1 week. Immediate availability and commitment to meet the timeline is essential. If you have the necessary skills and ex...
I am looking for a freelancer who can help me deploy Asterisk 16 with a specific PJSIP configuration. The ideal candidate should have experience with Asterisk and PJSIP. Requirements: - Familiarity with Asterisk 16 - Ability to configure PJSIP according to specific requirements - Experience in handling concurrent calls, with a focus on optimizing for a single call The requirements are very simple. I have configured it myself before and achieved single-pass. If the extension calls the mobile phone, the sound of the mobile phone can be heard, but the sound of the extension cannot be heard by the mobile phone. You only need to configure the phone to be able to achieve dual communication! If the price is not suitable, the price can be negotiated as long as you can solve...
Hosting a VoIP Softphone on a Windows Server Skills and experience required: - Strong knowledge and experience in hosting VoIP softphones on Windows servers - Proficiency in configuring and managing VoIP protocols, particularly SIP - Ability to handle concurrent calls ranging from 1 to 10 Project details: We are looking for a skilled freelancer who can help us host a VoIP softphone on a Windows server. The server will be running on the Windows operating system. Key requirements: 1. Operating System: Windows 2. VOIP Protocol: We are open to suggestions as the client did not specify a preference. 3. Concurrent Calls: The softphone should be able to handle 1 to 10 concurrent calls. If you have a strong background in hosting VoIP softphones on Windows servers, and are ...
...capabilities I need (Automatic Local Presence for Calls & SMS & a Power Dialer). So I'm trying to figure out a way to use both inside Salesforce, but am not sure what is possible. 1st idea is creating 2 separate open cti softphones within salesforce. One for all outbound calls (click to dial, dial pad, and power dialer) as well as all SMS features (360 CTI/SMS will have this part covered). The 2nd softphone (Vonage Business Center or possibly Vonage Contact Center) would handle all inbound calls so that I could have a ring first to & call tagging feature for the inbound call while still allowing a call tree/IVR/Virtual Receptionist to be in place. 2nd Idea is creating web hooks using either zapier or make when calls are made to the Vonage Desktop or Mobile A...
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
I want to add a VoIP client into our app to make and receive phone calls. Looking for iOS and Android developer To be able to send and receive SIP calls to our Freeswitch server I want to use something that doesn’t tie us to a CPAAS platform that requires us to buy phone numbers from them. Since we have our own VoIP sys...VoIP client into our app to make and receive phone calls. Looking for iOS and Android developer To be able to send and receive SIP calls to our Freeswitch server I want to use something that doesn’t tie us to a CPAAS platform that requires us to buy phone numbers from them. Since we have our own VoIP system .So something like pjsip -No need for registration- -no need for incoming calls- i just care about outbound calls for now Please note about your exp...
Reguired skills: I am looking for a skilled developer who can create a GenesysCloud React App with the following requirements: Specific Features: - Call Routing functionality is needed for the app. - Genesys Cloud SDK - Genesys WebRtc SDK Design/Layout: - The client has a specific design in mind for the app. Deadline: - The project needs to be completed within the next 1-2 weeks. Ideal Skills and Experience: - Strong proficiency in React and GenesysCloud development. - Experience with implementing Call Routing functionality. - Ability to work with specific design requirements. - Excellent time management skills to meet the tight deadline. If you have the necessary skills
OpenStage 20E SIP Firmware I am looking for a skilled professional who can assist me with installing the latest version of OpenStage 20E SIP Firmware. Requirements: - Experience with OpenStage 20E SIP Firmware installation - Familiarity with Asterisk PJSIP Specifics: - The current version of the firmware is not specified, so the freelancer should be able to handle any version (1.0, 2.0, or 3.0) - The main goal is to ensure compatibility with Asterisk PJSIP - The project requires immediate attention, so the freelancer should be available to start working on it right away Deliverables: - Successful installation of the new firmware - Verification of improved audio quality, enhanced security features, and bug fixes/stability improvements If you have the necessary skills and...
...Additional requirements: - Integration of Fail2Ban to enhance security and protect against unauthorized access - Implementation of PJSIP for improved audio quality and performance - Do not use FreePBX or any other Bloatware. Your solution will be rejected and you will not be paid if you do that. Has to be done only via asterisk. - Demonstrate working inbound voicemail service that plays a greeting and accepts voicemail on Telyx SIP lines. Ideal skills and experience: - Strong knowledge and experience with Asterisk and Ubuntu Docker - Familiarity with voicemail systems and call routing - Expertise in integrating Fail2Ban for security purposes - Proficiency in implementing PJSIP for enhanced audio quality - Asterisk has to run in Docker with a specific IP address in the s...
...Additional requirements: - Integration of Fail2Ban to enhance security and protect against unauthorized access - Implementation of PJSIP for improved audio quality and performance - Do not use FreePBX or any other Bloatware. Your solution will be rejected and you will not be paid if you do that. Has to be done only via asterisk. - Demonstrate working inbound voicemail service that plays a greeting and accepts voicemail on Telyx SIP lines. Ideal skills and experience: - Strong knowledge and experience with Asterisk and Ubuntu Docker - Familiarity with voicemail systems and call routing - Expertise in integrating Fail2Ban for security purposes - Proficiency in implementing PJSIP for enhanced audio quality - Asterisk has to run in Docker with a specific IP address in the s...
...Time Analytics Real Time Data Real Time Editing Real Time Monitoring Real Time Notifications Real Time Synchronization Real-Time Chat Real-time Updates Recording Reminders Remote Access/Control Remote Support Software Reporting & Statistics Reporting/Analytics Role-Based Permissions SSL Security Scheduling Screen Sharing Search/Filter Secure Data Storage Shareholder Management Single Sign On Softphone Software Status Tracking Survey/Poll Management Surveys & Feedback Task Management Task Planning Task Progress Tracking Task Scheduling Template Management Third Party Integrations Time Zone Tracking To-Do List Transcripts/Chat History Two-Factor Authentication Two-Way Audio & Video Unified Directory User Management Version Control Video Call Recording Video Chat Video C...
I am looking for someone who can help me investigate why my Softphone SIP Client is not connecting and trace the SIP packets. The ideal candidate for this project should have experience in SIP protocols and troubleshooting. Skills required: - Strong knowledge of SIP protocols and troubleshooting - Experience with Softphone SIP Clients - Familiarity with Windows operating system - Ability to trace and analyze SIP packets Responsibilities: - Investigate and identify the reasons why the Softphone SIP Client is not connecting - Trace and analyze SIP packets to identify any issues or errors - Work with the specific SIP provider to troubleshoot and resolve connectivity problems Please note that the Softphone SIP Client is running on a Windows operating system and ...
I am looking for someone who can help me investigate why my Softphone SIP Client is not connecting and trace the SIP packets. The ideal candidate for this project should have experience in SIP protocols and troubleshooting. Skills required: - Strong knowledge of SIP protocols and troubleshooting - Experience with Softphone SIP Clients - Familiarity with Windows operating system - Ability to trace and analyze SIP packets Responsibilities: - Investigate and identify the reasons why the Softphone SIP Client is not connecting - Trace and analyze SIP packets to identify any issues or errors - Work with the specific SIP provider to troubleshoot and resolve connectivity problems Please note that the Softphone SIP Client is running on a Windows operating system and ...
