Freeswitch voicexmlemplois

besoin d'avoir un script php(perl/lua) pour créer un bridge en freeswitch pour initialiser un appel. si on appelle le script avec un numéro de téléphone alors on crée un appel entre une extension (3000) et le numéro appelé

j'ai besoin de installer un serveur opensips avec une proxy rtp (rtpproxy) + serveur Freeswitch le tout avec debian jessie x64 comme sur le plan: Merci

Besoin des competneces pour des scripts lua avec freeSwitch et avec le mode SMS et chat

I need a specialist to resolve serious call routing issues in my telephony system based on FusionPBX and FreeSWITCH. Issues: - Calls not connecting - Incorrect call destinations - Dropped calls These problems emerged after a recent system upgrade.

I am looking for a C programmer who is also familiar with Freeswitch to patch up a bug that freeswitch themselves have not given any attention to. I found what I needed within the source code, it is just a matter of now making it work correctly. This is not anything complicated, should not take more than 2-3 hours at the most from my experience. Key Responsibilities: - Debugging existing C code within the Freeswitch framework - Identifying and resolving functionality bugs impacting call routing Ideal Skills: - Proficient in C programming - Familiar with Freeswitch - Strong debugging skills - Excellent problem-solving abilities

Hello: I need your help to - Install and configure Freeswitch on my Linux server (Debian) - Ensure the system is set up for optimal performance by using open source libraries Please only apply if you have a proven track record with similar projects.

I'm looking for a VoIP engineer who can implement a two-call method: the first call should ring for no more than half a ring, and the second call should go directly to voicemail. I've already set up this two-call method on FreeSWITCH, but I can't get the second call to not ring at the destination phone. I’d love to discuss how we can fix this. Key Responsibilities: - Implementing Ringless Voicemail on our existing system - Enhancing our customer feedback collection process - Ensuring optimal performance of the VoIP system Ideal Candidate: The ideal freelancer for this project would have extensive experience in VoIP, specifically with Ringless Voicemail. They should also have a deep understanding of customer service enhancement techniques and how to implement...

I need to relocate my existing FreeSWITCH setup to OpenSIPS + FreeSWITCH primarily for performance enhancements. Skills and Experience Needed: - Extensive experience with FreeSWITCH and OpenSIPS. - Strong understanding of voice over IP (VoIP) systems. - Proven track record in system migration. - Excellent problem-solving skills to ensure seamless transition with minimal downtime. - Ability to optimize the new setup for enhanced performance. 1 ) install opensips, opensips-cli, opensips-cp 1 h

...destination's WhatsApp number. - We will provide the phone number/phone numbers and pictures for the 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 2) WhatsApp gateway converts the SIP to ...

I'm looking for an expert to install FusionPBX on my Debian server. I have an existing PostgreSQL database server, so no need to set up a new one. However, I do need assistance with ...looking for an expert to install FusionPBX on my Debian server. I have an existing PostgreSQL database server, so no need to set up a new one. However, I do need assistance with the installation and initial setup. Key Requirements: - Installation of FusionPBX on a Debian server - Configuration of custom dial plans - Assistance with installation and initial setup - Set FusionPBX to use our fork of Freeswitch, not it's install instance Ideal Skills: - Extensive experience with FusionPBX - Proficiency in Debian - Experience with PostgreSQL - Ability to configure custom dial plans - Strong tr...

I'm seeking an expert to set up a FreeSwitch PBX-VOIP system on a Linux VPS for customer support purposes. Key Responsibilities: - Install and configure the FreeSwitch platform on a Linux VPS. - Prepare the server for optimal performance and security. - Implement the feature of voicemail to email. - Inbound & Outbound Calling Rules -Trunks setup Ideal Skills and Experience: - Extensive experience with FreeSwitch and Linux VPS. - Prior work with setting up PBX-VOIP systems. - Knowledgeable in server preparation and configuration. - Understanding of customer support communication needs. - Ability to implement and configure specific VOIP features.

I'm in need of a full-fledged Freeswitch Multi-Tenant UCaaS solution primarily tailored for business communication. This system should encompass a variety of features including VoIP calling, instant messaging, video conferencing, and more. Key Features: - VoIP calling - Instant messaging - Video Conferencing - API integration - Customer front-end GUI - Mobile and Desktop Apps -CRM itegration* -CCaaS -Call Reporting module *CRM Integration: Most softwares now allow API's or Webhooks, so I want the ability to utilize (write to those APIs) if that makes sense. I current have about 4000 extensions on my current system and some clients are linked to Salesforce, Hubspot, Zoho, SugarCRM. The solution needs to be robust enough to handle multi-tenancy. The ideal candidate shoul...

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.

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.

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...

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...

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

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...

...Expertise: Cloud Telephony Platforms: Proficiency in cloud telephony services such as Twilio, Plivo, or Nexmo, including API integration and service management. Implementation with CAAS platform like Genesys Connect and Avaya, VoIP Technologies: 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 bo...

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

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

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...

...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 ...

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.

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...

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.

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

Hi, We are looking for a reliable software development shop to help us maintain our B2B SaaS VoIP Telecom software system and add on new features. Key Requirements, programming languages in these languages is a must: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line MySQL Mongo DB React Electron GIT - GitHub Twilio Telecom knowledge FreeSWITCH OpenSIPs AWS Purpose: Our software is a B2B SaaS and will be focused on Telecom, CRM, Data, AI using Open AI API and Claude Sonnet API and Marketing. So, if you have experience in these fields, it would be a plus. Features: Also will need to handle customer data management, call tracking and analytics, marketing automation, reporting, and analytics. This will be a long-term partnership, so I'm keen to find a team that...

(unfortunately i only havebudget of $111 , due to last freelancer unable to complete and is already in balance) Freeswitch is already setup and running via whm root terminal, just need someone to install fusionpbx without interfering with currrent cpanel that has wordpress front end site on it. after completion will display via subdomain 1. freeswitch is running, no need help for this 2. just install fusionpbx for me on a sub domain, the sub domain is actually set to 7.4 3. cannot interfere with current installation of cpanel

(Dont overbid, should only take 30 min for experienced freelancer) Freeswitch is already setup and running via whm root terminal, just need someone to install fusionpbx without interfering with currrent cpanel that has wordpress front end site on it. after completion will display via subdomain 1. freeswitch is running, no need help for this 2. just install fusionpbx for me, pgsql database is already installed as well for fusionpbx i believe 3. cannot interfere with current installation of cpanel

I'm currently working on a project that requires expertise in FusionPBX configuration. I'm having trouble with the extension setup which needs immediate rectification. 1. SSL Certificate install 2. Configure Fail2Ban 3. Configure IPTables 4. Poly (Polycom) phones won't download config from server 5. Server will not issue config 6. Configure Shared Line Appearances/Multi...Configure Shared Line Appearances/Multiple Call Appearances 7. Configure SLA/MCA held calls to be able to be picked up on other phone 8. Configure SLA/MCA live calls to be barged in from other phones 9. Yealink phones do not connect 10. SIP Trunk disconnects from time-to-time 11. Create dialplan for phones so users don't have to press SEND or DIAL after typing in phone number You must be fluent in ...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

Preciso de um discador automático em freeswitch, segue algumas necessidades: Necessidades - Multitenant - várias empresas no mesmo servidor. - Franquias - Usuário com permissão de selecionar as suas franquias ou empresas cadastradas em seu nome, para que possa selecionar e visualizar todos os dados daquela empresa, todo o dashboard e relatórios, enfim, toda a interface da empresa. Possibilidade de colocar o logotipo de acordo com o domínio cadastrado. - Página administrativa para cadastrar um novo servidor freeswitch e setar qual a cps máxima desse servidor, para que possa aumentar ou reduzir a cps por servidor conforme necessário. - O sistema deve ser capaz de operar integralmente atrás do nginx, pa...

I need a professional to help me set up my FreeSWITCH instance with FusionPBX on my Linux dedicated server. The job requires some troubleshooting as I'm facing issues with my current setup and need a reliable, long-term solution. Key Points: - I have already installed FreeSWITCH on my server. But it's not working as expected - the socket keeps closing during voice tests. - I've checked the server requirements for FusionPBX, but I'm not sure if all the necessary dependencies and packages are properly installed. - FusionPBX seems to be the main problem - likely due to dependencies or configuration issues. Ideal Skills and Experience: - In-depth knowledge of FreeSWITCH and FusionPBX. - Proven experience in setting up and configuring VoIP solutions on...

I need help setting up a dial plan for my FusionPBX/FreeSWITCH system for emergency priority paging. Key Requirements: - Configuration: Experience in configuring FusionPBX and FreeSWITCH is essential. - Dial Plan: Specifically, you'll need to assist with setting up a dial plan that prioritizes emergency announcements. The focus of the system will be to deliver urgent messages in time-sensitive situations.

I'm in need of an experienced ASTPP/Freeswitch developer to create an Interactive Voice Response (IVR) system for my project. The primary goal of this IVR system is to Play a pre-recorded message when a customer calls a DID, then route the call according to any routing rules configured on the dialplan. This means that the system needs to be designed to handle customer queries and direct calls to the appropriate department. Ideal Skills: - Proficient in ASTPP/Freeswitch - Experience in IVR system development - Strong understanding of customer support processes - Ability to design system for call routing The project is expected to be completed within a reasonable timeframe. A successful candidate will be able to understand the nuanced requirements of the system and d...

I am seeking a highly skilled freelancer with intensive experience in FusionPBX and Freeswitch. The main objective of this project is the creation of a precise dial plan for FusionPBX. The developed dial plan should effectively handle active calls - holding them temporarily during paging calls. Key tasks include: * Dial plan must allow higher priority calls to override held calls * Acceptance of priority levels for different callers as input parameters The ideal candidate should possess these skills and experience: * Proven experience in FusionPBX and Freeswitch * Knowledge of implementing dial plans * Understanding of call flows and priority handling. Your task will revolve around ensuring a smooth call flow process during the paging, which detailed understanding of ...

I am in need of an expert OpenSIPS VOIP Developer who can help create a robust system with high quality call routing. The main feature's to be implemented are NAT Handling in the Cloud, Microsoft Direct Routing, and Mid Registrar. The system should be designed to interact efficiently with PBX systems (FreeSWITCH) and OpenSIPS. Required Skills: - Extensive experience with OpenSIPS VOIP system development - Strong understanding of direct call routing - Proficiency in integrating VOIP systems with PBX and OpenSIPS - Familiarity with general telecoms concepts and technologies. The ideal freelancer should have worked on similar projects and be ready to showcase relevant examples in their proposal. If this project goes well, there could be more ongoing work. I look forward to disc...

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

I require an expert in Asterisk and FreeSWITCH development, along with experienced system engineering skills. Specifically, I need help with: - Asterisk and FreeSWITCH development - Integrating VoIP - Setting up Call routing and IVR (Interactive Voice Response) - CTI - Skill group base call routing to agents. Apart from these, the implementation of Asterisk and FreeSWITCH clustering as well as Call Center Reporting is required. The technology stack should be Linux, PHP, MySQL along with postgresql and API knowledge. The freelancer should have considerable experience in these to meet the project's specific requirements.

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

I'm looking for an OpenSIPS expert with comprehensive knowledge in call routing. currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have iss... currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have issued when A call B, and B ring 180, after that B refer call to C, call not reach to C - I'm follow opensips Docs we have to config uac_redirect but i try with no luck 2. We use media server is Freeswitch, how can i pass a variable to Freeswitch, i try add header follow format: X-Variable, sip_h_X-Variable but in luascript of FS...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related applications

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

I have a particular project requiring expertise in Freeswitch, preferably with exposure to a Debian Linux environment. The key objective is to use Freeswitch for answering machine detection. The main requirement is to capture and analyze voicemails. We want to build a custom module or ready solution which can detect human or machine on the outbound dialer. The solution should work on any freeswitch version and without any licensing restriction. Please note, we have already used AVMD module and it just detects beep. We want to detect the machine or human from the initial screening of voice only. Skills and Experience Needed: - Solid experience with Freeswitch answering machine detection - Familiarity with working on Linux systems - Ability to effectively capt...

Task details - For ISUP/SS7 (E1 card implementation) along with IVR knowledge in any open source SIP servers like Freeswitch/Yate/Mobicents etc. or working experience in CRBT server. Interested candidates please apply