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2,000 freeswitch voicexml asterisk 見つかった仕事

I'm looking to implement a system on my Asterisk server that checks a caller's credit based on the number dialed. Key Requirements: When user tries to make a call, send the dialled number to the API, the API will return how many minutes of credit they have for that number. If they have > 1 minute of credit available then allow the call to go ahead, otherwise send caller to a pre-recorded message advising them they have no credit (we will supply the message). During the call if credit falls below 2 minutes play "beep" warning to the caller every 15 seconds, when credit runs out disconnect called party and send caller to pre-recorded message advising they ran out of credit (we will supply the message). The API is a standard RESTful API that works with JSON tha...

$184 Average bid
$184 平均入札額
37 入札

I need a developer to create a web-based softphone for me using Asterisk, WebRTC, and React. This softphone should be able to support call recording as a key feature. We regard this as quite an easy project for a developed with experience working with Asterisk, WebRTC. The softphone will be deployed at our website.

$545 Average bid
$545 平均入札額
41 入札

I'm looking for an experienced developer to configure a SIP soft on my ViciDial/Asterisk setup. The autodialer should utilize Python and PJSIP, and potentially incorporate existing autodialer code. Key project specifics include: - PJSIP as the preferred autodialer software - Implementing functionalities for outbound calling and playing pre-recorded files - Incorporating Google Text-to-Speech for call content - Setting up an auto dialing mode that pulls data from a database Ideal candidates should have a strong background in Python, PJSIP, and Asterisk, with prior experience in setting up autodialers. Knowledge of Google Text-to-Speech and database integration will be highly advantageous.

$7 / hr Average bid
$7 / hr 平均入札額
2 入札

Asterisk & WebRTC Installation on Linux

$145 Average bid
$145 平均入札額
24 入札

I'm looking for a skilled developer with experience in FreeSWITCH to create a custom phone system for me. The primary functionality needed is the development of a comprehensive phone system. Key features of this phone system will include an automated attendant. Ideal skills for this project include: - Proficiency in FreeSWITCH - Experience in developing VoIP systems - Knowledge of automated attendant systems - Call routing programming skills - Voicemail system development experience Please include examples of similar projects you've worked on in your proposal.

$2763 Average bid
$2763 平均入札額
34 入札

I'm looking for a skilled Freeswitch developer to create a robust conference calling platform. Key Features: - The platform must support more than 100 participants simultaneously - Essential VoIP feature: Call recording Ideal Skills and Experience: - Proven experience with Freeswitch - Expertise in developing VoIP services, particularly conference calling platforms - Knowledge and experience in implementing call recording features - Ability to create scalable and reliable systems Please provide examples of similar projects you've completed in your proposal.

$93 Average bid
$93 平均入札額
3 入札

Busco experto para configurar un server de asterisk y diferentes modulos con mms sms entre otras

$174 Average bid
$174 平均入札額
14 入札

Job Title: Core Java Developer Brief: Kirusa, Inc a leading company that provides many Value Added Services and AI based solutions to Telcom Companies in more than 18 countries world wide is looking fo...communication skills. Educational Qualifications: Bachelor’s degree in Computer Science, Information Technology, or a related field. Preferred Qualifications: Experience in using cloud based deployment solutions Experience with RESTful APIs and web services. Knowledge of front-end technologies is a plus. Familiarity with Agile development methodologies. Experience with VoiceXML based application development. Benefits: Competitive Compensation Professional development opportunities Flexible working hours Friendly and supportive wo...

$1446 Average bid
$1446 平均入札額
27 入札

I'm looking for a skilled developer to create a softphone for me using Asterisk, WebRTC, and React. The softphone should support the following: - Functionality: The softphone must be capable of handling voice calls. - User Authentication: The softphone does not require any form of user authentication. - Platform Compatibility: The softphone should be designed to work exclusively on desktop platforms. Ideal skills for this project include: - Extensive knowledge and experience with Asterisk, WebRTC, and React. - Proven track record of developing desktop softphones. - Ability to design a user-friendly, functional, and efficient softphone. - Experience in creating software with no user authentication requirements. Please submit your proposal if you meet these criteri...

$488 Average bid
$488 平均入札額
25 入札

Estoy en busca de una persona con amplio conocimiento en Asterisk para que me apoye en la revisión y configuración de un servidor con FreePBX y Asterisk. Actualmente, estoy intentando establecer una conexión adecuada para realizar llamadas desde un CRM propio mediante AMI, pero no he logrado configurar correctamente la conexión, a pesar de haber intentado varias alternativas. Si tienes experiencia en este campo y estás interesado en colaborar, por favor, contáctame para brindarte más detalles en privado.

$214 Average bid
$214 平均入札額
11 入札

...integration will be with an Asterisk server through the Asterisk Manager Interface (AMI). I need the following functionalities from the VoIP integration: - Call logging: I need all calls to be logged for future reference. - Click-to-call: This feature should allow me to call any number with a single click. - Call routing: I want to be able to route calls to different extensions. Additionally, I require some customer interaction features: - Voicemail management: I need help managing voicemails. - SMS Integration: I want to be able to send and receive SMS through the system. - Call Analytics: This feature should provide me with insights about the calls. The ideal freelancer for this project should have extensive experience in VoIP services, particularly Asterisk,...

$60 Average bid
$60 平均入札額
1 入札

I am seeking an expert in Asterisk server setup. The primary objective is to configure the server for outbound marketing campaigns. Key Responsibilities: - Set up the Asterisk server to facilitate outbound marketing calls. - Integrate the server with a PHP-based CRM system for seamless operations. Ideal candidates should have: - Extensive experience with Asterisk server configuration. - Proficiency in integrating Asterisk with CRM systems, particularly PHP-based ones. - A solid understanding of outbound marketing call requirements. - Excellent problem-solving skills and attention to detail. Please ensure your bid reflects your capabilities and previous experience with similar projects. Thank you.

$195 Average bid
$195 平均入札額
19 入札

I'm looking for an expert to install a FreeSWITCH + ASTPP (or you can offer other option here) VoIP SIP management platform on my Debian/Ubuntu server. The main goal is to manage SIP users and bill them efficiently through web interface. Key Features: - User Management: the platform should allow management of SIP users (accounts) - create, edit, delete, assign them to groups - Rate management: the platform should manage call rates according to call directions - Support of different vendors and its configuration (priopity and specific settings to choose specific vendor), call costs - Billing: the platform needs to accommodate postpaid billing, real-time usage tracking, and support for pre-paid voucher cards. - Provide internal calls capability (between system users) - Repo...

$550 Average bid
$550 平均入札額
33 入札

I'm looking for an experienced freelancer who can assist with Asterisk, PHP, Issabel PBX, and FreePBX development and configuration. Key responsibilities include: - Configuring Issabel PBX and FreePBX to meet our requirements - Implementing custom PHP applications - Setting up Asterisk according to our specifications Your application should focus on showcasing your experience with these technologies. Feel free to provide any relevant samples of your past work. I expect the successful candidate to have an intermediate level of expertise with the aforementioned technologies.

$13 / hr Average bid
$13 / hr 平均入札額
12 入札

FusionPBX/FreeSwitch Advanced configuration, including multiple call appearances/shared call appearances 1. SSL Certificate install Tell me what type of free certificate I need to supply to you. 2. Configure Fail2Ban Exclude the following IPs from getting blocked ever. 3.229.25.209/32 3. Configure IPTables Verify the above IPs will NEVER get blocked. For the following tasks, in addition to any instructions listed, you may need to do additional coding or configuration. The task will not be considered complete unless the feature works properly in real world. 4. Configure Shared Line

$544 Average bid
$544 平均入札額
11 入札

I'm in search of a proficient developer to help deploy a FreeSwitch-based call center solution. The project involves setting up a call center solution based on FreeSwitch, a popular open-source platform with extensive features for handling calls. Below are some of the key features FreeSWITCH offers for call centers: 1. Advanced Call Routing Skills-Based Routing: Directs calls to agents based on their skills, ensuring that customers are connected to the most qualified agent. Time-Based Routing: Routes calls based on the time of day, allowing for different handling during business hours, after-hours, and holidays. Geographical Routing: Routes calls based on the caller’s location for region-specific service. 2. Interactive Voice Response (IVR) Customiza...

$1120 Average bid
$1120 平均入札額
38 入札

...color preferences and branding. Skills and experience needed for this project include: - Graphic design - Creativity in designing visually appealing covers - Understanding of modern and clean design aesthetics - Ability to incorporate branding elements, if provided - Strong communication skills to understand and interpret my vision for the cover My name: Selena Beaumont Agencies names with an asterisk are the strong considerations: The Briar Agency* Thistle Literary* Solstice Literary The Hestia Agency* Neith Media* Fallow Media* Hearth Literary* Dark green has been a color consideration. This is a Creative Media agency for Authors, Podcasters, Influencers, Speakers, and TV & Film professionals. NO stock photos of people/stock photos needed - focus purely on typography...

$25 Average bid
保証
$25
191 エントリー

...Understanding the fundamental principles of VoIP, including how voice data is transmitted over IP networks. Familiarity with VoIP software like Asterisk, FreeSWITCH, and Cisco Call Manager is crucial. SIP Protocol: Proficiency in the Session Initiation Protocol (SIP), which is essential for setting up and managing VoIP communications. Skills in configuring SIP trunks, managing SIP sessions, and using SIP tools such as Wireshark for troubleshooting are important. PBX Systems: Knowledge of Private Branch Exchange (PBX) systems, including both traditional hardware-based PBX and modern IP-PBX systems. Experience with popular PBX platforms like Asterisk, 3CX, and Avaya is valuable. Network Protocols: Understanding networking protocols such as TCP/IP, UDP, and RTP, which ...

$21 / hr Average bid
$21 / hr 平均入札額
13 入札

I'm in need of a skilled PHP developer with experience in FusionPBX. Yes fusionpbx and freeswitch is running and installed I have an existing PHP script that communicates with FusionPBX, but I'm encountering some issues. Ai created the connection but i think it created it wrong, examine the attach script and just fix it so that it can correctly connect to my fusionpbx software on my ubuntu server. i am using this within a custom wordpress plugin ive provided the logic just need an experienced person that can make it work

$26 Average bid
$26 平均入札額
10 入札

I'm in need of a skilled PHP developer with experience in FusionPBX. Yes fusionpbx and freeswitch is running and installed I have an existing PHP script that communicates with FusionPBX, but I'm encountering some issues. Ai created the connection but i think it created it wrong, examine the attach script and just fix it so that it can correctly connect to my fusionpbx software on my ubuntu server. i am using this within a custom wordpress plugin ive provided the logic just need an experienced person that can make it work

$17 Average bid
$17 平均入札額
12 入札

I need a custom switch developed, similar to Veriswitch, with a focus on cost-efficient call routing. The switch should incorporate Must Use OpenSip Not Freeswitch: - Call Routing Based on Cost: The switch should be capable of identifying the least cost route for calls. - Real-Time Monitoring: The ability to monitor the performance and status of calls in real time. - Failover and Redundancy: Ensuring continuous connectivity and minimizing downtime by switching to backup routes or servers when necessary. The frontend of the switch should be built using Bootstrap and HTML with PHP. The switch primarily needs to handle: - VoIP Calls: The switch should be compatible with Voice over Internet Protocol calls. - SIP Trunking: It should also support Session Initiation Protocol tr...

$1339 Average bid
緊急
$1339 平均入札額
44 入札

I am looking for a proficient Asterisk specialist to connect Asterisk with a Trunk Gateway using SIP. The main goal is to enable the system to make and receive calls via the Gateway. Key Objectives: - Establish a seamless connection between Asterisk and the Trunk Gateway using SIP. - Ensure the system is fully capable of making and receiving phone calls. Ideal Skills: - Strong experience with Asterisk and SIP. - Proven track record of successful integrations with Trunk Gateways.

$34 Average bid
$34 平均入札額
4 入札

Job Title: Asterisk and Node.js Developer for Real-Time Call Transcription and Summary Service Description: We are looking for a highly skilled Asterisk and Node.js developer to create a real-time call transcription service. This service will transcribe calls using ChatGPT's speech-to-text functionality, summarize the transcription, store the summary in a database, and send an SMS to the user with the summary. The project duration is a maximum of 3 days. Key Responsibilities: Integration with Asterisk: Configure Asterisk to route calls to a Node.js service. Utilize Asterisk's ARI (Asterisk REST Interface) to manage calls. Real-Time Transcription: Implement real-time call audio capture. Use ChatGPT's speech-to-text API to transcribe th...

$440 Average bid
$440 平均入札額
129 入札

...of an expert who can fix two separate issues I am experiencing with my FreePBX instances. - On the first FreePBX instance, I am facing an issue where I cannot connect to Asterisk. This seems to be causing the system to not make any outgoing calls. In addition, I cannot access outbound trunk, extension, outbound calls,. This is the primary issue that needs urgent attention. - For the second FreePBX instance, I am encountering a PHP warning unable to load the dynamic library for various modules. This causes GUI to be blank page. To be successful in this project, you should have: - Proven experience in FreePBX, Asterisk, and PHP - Deep understanding of VoIP systems - Ability to troubleshoot and resolve issues efficiently Your role is to identify the root cause of these...

$18 / hr Average bid
$18 / hr 平均入札額
49 入札

...America , low internet speed , we need use vpn installed on their computers I'm seeking a tech-savvy freelancer to assist with managing and providing support for my Asterisk VICIdial VOIP system. The main issue we're currently facing is system crashes or freezes. Key responsibilities include: - Troubleshooting and resolving system crashes or freezes promptly. - Managing call routing and monitoring processes within the system. - Overseeing user management and ensuring system security. Ideal skills and experience: - Proficiency in Linux administration to efficiently handle system issues. - Strong knowledge in Asterisk configuration to optimize system performance. - Proven experience in VOIP troubleshooting to address any system glitches. If you're ...

$20 / hr Average bid
$20 / hr 平均入札額
27 入札
$181 平均入札額
24 入札

Proficient asterisk expert help edit voice group call if you are not Pro with asterisk dont use time here

$56 / hr Average bid
$56 / hr 平均入札額
18 入札

...the call to the destination's WhatsApp number. - We will provide the phone number/phone numbers WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with multi-phone number or can use just one WhatsApp account with multi calls. ) - The development platform/operating system is not important. you can use Asterix, Freeswitch, FreePBX vs... - The implementation should return the correct call error codes to the SIP backend like CALL SUCCESS(200 OK), BUSY(486 Busy Here), UNAVAILABLE(503 Service Unavailable), etc.... to try other rounds on the other SIP Switch. Functional flow 1) Calls from PBX/sip gateway will be sent to WhatsApp gateway (phones) 2) WhatsApp gateway gateway (phones) converts ...

$184 Average bid
$184 平均入札額
17 入札

I'm looking for a professional who can set up a...engine. The main purpose of this feature is to read out scripted messages. This will be useful for our communication goals. You should have experience in: - Setting up outbound dialer systems - Integrating softphone applications with APIs - Customizing dialer systems to integrate text-to-speech engines - Want api or db interface for getting the status of calls made - Experience with OpenPBX, Asterisk or other platform will be plus it will be of added advantage if it can be linked with CRM software like VCDial so agent can be connected on user response so user can speak to them. i can provide necessary hardware and setup and need a software which i can use easily for multiple PRi line. Will need help and training so I can mai...

$314 Average bid
$314 平均入札額
16 入札

I need a phone validation system that can identify out-of-service numbers and integrate with Asterisk. The system should be able to: - Validate phone numbers. - Identify out-of-service phone numbers. - Integrate seamlessly with Asterisk. Ideal skills for this project include: - Proficiency in Asterisk. - Experience with phone number validation. - Knowledge of database management (Mysql) - Strong communication skills for clear updates and collaboration.

$154 Average bid
$154 平均入札額
41 入札

I have a Linux server and need Asterisk, an open-source PBX software, installed on it. Key Requirements: - Asterisk Installation: The latest stable version of Asterisk needs to be properly installed on my Linux server. - Configuration: After the installation, basic configuration settings should be set up for smooth operation. Ideal Skills: - Proficiency in Linux: You should be experienced in working with Linux servers. - Asterisk Installation Expertise: Prior experience installing Asterisk is highly appreciated. Note: Please make sure you ask about root access and need for a specific version of Asterisk during the clarification phase.

$32 Average bid
$32 平均入札額
4 入札

I'm looking for a skilled professional who can help me implement and configure an Asterisk server for call center functionality. Key requirements: - Implementation and configuration of an Asterisk server from scratch (as we do not currently have it installed). - Prior experience in setting up call center functionality is a must. The features that are most important to our call center functionality include: - Automated call distribution (ACD) - Interactive voice response (IVR) - Call recording Your past experience with similar implementations, as well as your understanding of VoIP technologies, will be crucial. Please provide details of your relevant experience in your proposal.

$785 Average bid
$785 平均入札額
54 入札

Project Overview: I am seeking an Asterisk PBX developer who can fully customize our phone system to best suit our needs. The ideal freelancer for this job should be an expert in: Custom configuration of Asterisk PBX VoIP trunking setup Integrating third-party systems such as CRM with Asterisk PBX We are looking for an experienced VoIP engineer to set up an Asterisk PBX system integrated with a UC2000-VG-32T GSM/5G gateway. The UC2000-VG-32T will be configured by the supplier, and the freelancer will need to coordinate with the supplier to ensure seamless integration and functionality with the Asterisk PBX system. The setup will support 100 local agents and 250 remote agents, handling both voice calls and SMS. Key Tasks and Features: One of the key task...

$35 / hr Average bid
$35 / hr 平均入札額
13 入札

I'm seeking an expert to provide step-by-step instructions on installing FreeSWITCH PBX on AWS. All the documentation that I have found online has failed.

$173 Average bid
$173 平均入札額
20 入札

I've a Jio mobile number and I want it to receive incoming calls via Asterisk on my shared hosting. This setup should include IVR and PBX functionalities. Furthermore, I'd like it to integrate with mobile apps like Zoiper, Linphone, and Bria. Further, I would like the complete system to run on zero budget without any third-party paid services. To be considered for this project, freelancers should have relevant experience with Asterisk and integrating it with mobile applications. A successful application will demonstrate: - Previous experience with Asterisk installations and configurations - Experience with setting up IVR and PBX systems - Familiarity with mobile app integration, particularly with Zoiper, Linphone, and Bria The main goal for this setup is t...

$133 Average bid
$133 平均入札額
12 入札

I'm looking for a freelancer with experience in Asterisk PBX systems to help me with the following key tasks: - Setting up Voicemail to Email - Configuring Call Forwarding - Implementing IVR (Interactive Voice Response) - Setting up Extensions - Configuring Trunking For your application, I'd appreciate if you could share any relevant past work you've done with Asterisk PBX systems. Looking forward to your proposals.

$100 Average bid
$100 平均入札額
10 入札

Possuo um hardware e gostaria de instalar / configurar uma distro para fazer a discadora rodar em meu Callcenter.

$200 Average bid
$200 平均入札額
8 入札

I need a skilled Freeswitch developer to enhance the functionality of the eavesdrop feature The current function of eavesdrop is: - Admin calls a feature code that contains the extension number of the user they want to monitor - Admin can listen to the phone call - When the remote extension hangs up the call, the admin is also hung up on - I read across forums that you can somehow attach the eavesdrop function to a conference bridge in order for the admin to be able to stay on the eavesdrop feature code for several hours. The conference bridge will then "activate" itself whenever the monitored user places or receives a phone call Key Details: - **Current Problem:** Admin gets hung up on randomly while eavesdropping. - **Desired Change:** Admin should be able to eavesdr...

$516 Average bid
$516 平均入札額
23 入札

I need help configuring a specific version of Asterisk PBX on AWS. I have the Airtel Configuration details for the setup. The task involves: - Setting up Asterisk PBX on AWS - Integrating Airtel Configuration details - Connecting the configured system with my Lead Dial CRM application also hosted on AWS Ideal Skills: - Proficiency in Asterisk PBX configuration - Experience working with AWS - Knowledge of CRM application integration - Familiarity with Airtel Configuration Please only apply if you have all the necessary skills and experience.

$59 Average bid
$59 平均入札額
8 入札

I have an existing FreeSwitch API project built with GoLang, designed for voice communications. I'm seeking a skilled freelancer to assist with deploying this project on a Debian 11 server. Key Responsibilities: - Utilize strong server-side knowledge to deploy the project effectively - Ensure the deployment is successful and the API is fully functional Requirements: - Proven experience with similar deployments - Strong understanding of GoLang, FreeSwitch, and Debian 11 - Ability to troubleshoot and provide solutions to any issues that may arise during deployment - Excellent communication skills and a proactive approach to problem-solving When applying for this role, please include examples of your past work with similar deployments.

$100 Average bid
$100 平均入札額
4 入札

I need a skilled web developer to create a responsive HTML templat...but for now, i just need html, css key points and attached files: 1. explore page = ( home page) explore 2. profile page = p360 & p360 (for the profile there is a tab at the top. i provided 2 png, but its all one page just showinghow both tabs loook in profile active and inactive) 3. discussion page = , (explanation) my server has freeswitch and fusionpbx meaning im able to create radio channels, this page will allow for open radio type chatroom, via voice and typing. yes the mute button and mic is an icon, with inactive and active function the wave will respond to noise once again, all of this is html template that i will be inserting into wordpress child theme

$160 Average bid
$160 平均入札額
202 入札

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all det...pay clients for their incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" includin...

$530 Average bid
$530 平均入札額
30 入札

I'm looking for an expert in OPENSIP and Asterisk to work on a high-traffic, scalable outbound and inbound call system. Key responsibilities include: - Setting up OPENSIP for SIP traffic management - Implementing call routing based on media server load - Developing IVR (Interactive Voice Response) features - Integrating Asterisk as the media server with the OPENSIP setup - Initiate call after reading the relevant data from Kafka through opensips and route calls to asterisk only for media. - Initial setup will have 1 opensips server and 5 media servers. The ideal freelancer for this project should have: - Extensive experience with OPENSIP and Asterisk - Proven track record with setting up highly scalable call systems - Strong understanding of IVR developm...

$326 Average bid
$326 平均入札額
9 入札

I'm in need of a skilled individual who can assist me in the deployment and configuration of a Free PBX (Asterisk) on an existing Linux based Cloud Server. The aim is to set up a functional PBX system with the following features: - Creation of Softphone Extensions: I will require the creation of softphone extensions that will be utilized for communication within the organization. - Trunk Lines Configuration: These lines are to be configured for both inbound and outbound calls. Your expertise will be required in ensuring that these lines are properly set up and operating smoothly over IP. - Operating Procedures and Policies: Setting up operating rules and policies that will govern the use of the PBX system. This includes the implementation of call forwarding and routing accor...

$121 Average bid
$121 平均入札額
18 入札

I'm looking for a skilled designer to create a display stand that's focused on showcasing small-sized tow balls and ...to have 5 towballs on one line of red dots at front, 5 towballs behind on those red dots, 4 bigger towballs behind this line on red dots and 4 bigger towballs on the next line on red dots – then remove red dots) please see tow ball positioning image to see where towballs are to be placed - In the hole in the front would be a 70mm matt black towball mounted - On the designed display on the red asterisk please put Australian made logo (attached) in top left hand corner and Holroyd logo (attached) in the top right hand corner - both need to be in small sizing - We will require this finished with bleed lines in and AI, PDF and EPS format in high res...

$67 Average bid
保証

Description We are seeking a highly skilled VoIP expert to assist us in integrating Asterisk (FreePBX) and Janus into our Angular web application. please have a extensive experience in setting up and configuring FreePBX and Janus, as well as integrating these systems into a web application environment. Project Scope: Setup FreePBX: Install and configure FreePBX on our server. Ensure proper configuration for VoIP functionalities. Set up necessary SIP trunks, extensions, and routes. Setup Janus: Install and configure Janus WebRTC Server. Ensure secure and efficient communication between Janus and FreePBX. Set up necessary plugins and configurations for optimal performance. Web App Integration: Integrate Janus with our Angular web application. Implement real-time communication fea...

$459 Average bid
$459 平均入札額
23 入札

I am looking for a skilled developer who can create a simple tool that can convert all incoming calls to SIP protocal g729 codec. You will advise the choice of your OS Key requirements: - The tool needs to conver...all incoming calls to SIP protocal g729 codec. You will advise the choice of your OS Key requirements: - The tool needs to convert a variety of all existing voice call protocols including H.323 and 232 ITU-T to SIP g729. - I don't have an existing Asterisk PBX system, so this tool should be standalone and not dependent on an existing system. Ideal Skills: - Proficiency in Asterisk, (Opertating systems such as CentOS, Ubuntu etc), and SIP protocol is a must. - Experience with protocol conversion tools iss a must. - Simple standalone Asterisk P...

$22 Average bid
$22 平均入札額
4 入札

I'm experiencing poor call quality on my Asterisk server, which is deployed on a Wide Area Network (WAN). The specific issue seems to be linked to the voice transmission codec, currently set to G.711. Key aspects of the project include: - Investigating and identifying the root cause of the poor call quality - Suggesting and implementing solutions to enhance the call quality - Potentially changing the codec from G.711 to a more suitable one, if necessary Ideal skills and experience for this project include: - Proficiency in Asterisk servers and VoIP technologies - Strong understanding of network configurations and troubleshooting - Knowledge of different voice codecs and their impact on call quality - Experience in improving call quality on WANs Your expertise and expe...

$98 Average bid
$98 平均入札額
7 入札

I'm experiencing poor call quality on my Asterisk server, which is deployed on a Wide Area Network (WAN). The specific issue seems to be linked to the voice transmission codec, currently set to G.711. Key aspects of the project include: - Investigating and identifying the root cause of the poor call quality - Suggesting and implementing solutions to enhance the call quality - Potentially changing the codec from G.711 to a more suitable one, if necessary Ideal skills and experience for this project include: - Proficiency in Asterisk servers and VoIP technologies - Strong understanding of network configurations and troubleshooting - Knowledge of different voice codecs and their impact on call quality - Experience in improving call quality on WANs Your expertise and expe...

$112 Average bid
緊急
$112 平均入札額
5 入札