Freeswitch voicexmlPekerjaan
We request the customization of the linphone app for the Android, iOS, and Windows platforms, incorporating our logo and company informatio...receive SMS text messages through the ASTPP PBX server Implementation of secure communications through zRTP, TLS, and SRTP Support for Bluetooth headset devices Language options: English, Chinese Integration with ASTPP PBX server to check balance Integration with ASTPP user accounts Message Archive Management (MAM) functionality Implementation of PUSH Notification from the ASTPP PBX server (FreeSwitch) Customization of the app with our logo and company information Provision of source codes. Kindly consider these requirements for the customized linephone app for the Android, iOS, and Windows platforms, and integration with our ASTPP P...
...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 WhatsApp p...
I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables server 1 : kamilio + LCR module + LCR module stored in Mysql tables Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
I am looking...the freelancer to determine the best configuration. Regarding the authentication method for the SIP server/client, I am not sure if a secure authentication method is required. Further clarification on this matter would be appreciated. Ideal Skills and Experience: - Experience with configuring Zenitel tcis3 system - Familiarity with Windows-based SIP server/client software, such as Asterisk, FreeSWITCH, or 3CX - Strong knowledge of SIP protocols and configurations - Ability to provide recommendations and suggestions based on project requirements The systems will be for a internal/private network. So should not be reliable on a internet connection. So no cloud solutions. This is a basic setup so tcis3 can call windows pc and Windows pc can call tcis3. Nothing els...
ng-voice, a trailblazer in the realm of open-source and VoIP technology, is currently seeking experienced freelance software developers to participate in various cutting-edge projects. Our cloud-native VoLTE/IMS solutions allow mobile network operators to ...execution. Required Skills and Experience: Strong interest in and understanding of telephony. Proficiency in C programming language Experience with signaling standards such as SIP, DIAMETER is a plus. Fluency in English; reading 3GPP specs should be a part of your skillset. Proven track record in adhering to deadlines and keeping projects on track Our Technology Stack Includes: Kamailio, RTPEngine, FreeSwitch Redis, MySQL, and RabbitMQ Docker; experience with Docker Swarm and Kubernetes is beneficial SonarCloud, BugSnag SIP, RTP...
We are looking to hire an experienced Full Stack Developer (7-8 years of experience in Node.js with Freeswitch on an immediate basis to help us keep growing. This position is full-time and can be done remotely. Requirements Strong organizational and project management skills. Proficiency with fundamental front-end languages such as HTML, CSS, and JavaScript or React.js Proficiency with server-side languages such as Node.js Experience working with Freeswitch Familiarity with database technology such as MySQL, and/or MongoDB. Excellent verbal communication skills. Good problem-solving skills. Attention to detail.
...providing an API-like interface to queue RVM drops and to be able to push CDRs based on webhook settings set on JSON request body. It should handle an audio file per request and be able to have multi-carrier support. It could keep a database for CDRs, configurations and eventually requests log for auditing. As this is for high volume, it should be developed on a compiled language like Go and use Freeswitch dialer for scaling communications stack (programming language and SIP stack could be discussed). It is required extensive experience working with Ringless Voicemails on a high performance scenario and with verified high drop ratio, as well as experience working with SIP communications. This is a backend project, no UI is required and it will work as a module of a bigger ...
We need a skilled freeswitch or asterisk developer to help develop an API interface capable of initiating phone calls on freeswitch (similar to Twilio APIs for voice calls). Sample use cases are to embed the API in a CRM or webservice to enable to user click a button on an external web application, and initiate a call on the freeswitch machine and the call is then connected to a softphone or webrtc endpoint for audio.
...looking for a Kamailio Developer! Our in-house Kamailio developer is retiring. We would love to find someone who is a real Kamailio expert that loves creating/developing outside-of-the-box solutions. We want to find someone who has experience with custom modules, and who has a fantastic fine-tuning skill set. All our Kamailio are extremely high CPS it would be ideal if you have experience in Freeswitch & Postgress as well. This is not a one-project job and done, we would love to find someone for the long term and become our "Go to Guy" and have someone get familiarized with our network/systems. The first task we have is for someone to review one of our new configurations and see where we can fine-tune it, also want to change how Kamailio communicates to the cus...
...looking for a Kamailio Developer! Our in-house Kamailio developer is retiring. We would love to find someone who is a real Kamailio expert that loves creating/developing outside-of-the-box solutions. We want to find someone who has experience with custom modules, and who has a fantastic fine-tuning skill set. All our Kamailio are extremely high CPS it would be ideal if you have experience in Freeswitch & Postgress as well. This is not a one-project job and done, we would love to find someone for the long term and become our "Go to Guy" and have someone get familiarized with our network/systems. The first task we have is for someone to review one of our new configurations and see where we can fine-tune it, also want to change how Kamailio communicates to the cus...
I am looking for a freelance IOS developer to create a SIP phone app for my existing business. The app needs to connect to my FusionPBX server, which is already set up. My ideal candidate would have experience in developing IOS SIP phone apps and be familiar with the FusionPBX system. The user interface customization required is moderate. I would like the app to...customization required is moderate. I would like the app to have shared call appearances feature and a clean, user-friendly design. Key requirements: - IOS SIP phone app development experience - Familiarity with FusionPBX system - Moderate level of user interface customization - SHARED CALL APPERANCES - MULTIPLE REGISTRATIONS/EXTENSIONS HANDLED IN THE APP - Clean and user-friendly design. - FREESWITCH AND FUSION PBX E...
This would be to develop and install a small call flow that runs on FreeSWITCH to screen calls. YOU Must know FreeSWITCH and have used it. You will need to install the FreeSWITCH stack and then write custom call flow for it. Integrates with Google Contacts API and Pushover notifications. you should have experience building call flows, databases, webforms/pages etc... this call flow will answer the call if the caller is in the subscribers google contacts # list, you will send the call to a second # if they are NOT in the google contacts # list, you will ask the caller to say their name and company name you will record that and transcribe it you will send the transcription via pushover to the subscriber you will the call the subscriber at a different number and play ...
I am seeking a FreeSWITCH or Asterisk expert to help me set up a new enterprise. The ideal freelancer should have prior experience in this area. The project requires call routing and forwarding features to be implemented. Other functionalities such as IVR and voicemail, as well as conference calling and recording may also be required. Please provide your experience in this field in your application. Automation will be required for this as well. I basically want people to go to my website, sign up, and get started with processing calls. Thank you.
I am looking for a Flutter developer who can help me connect my app with FreeSWITCH via SIP. Specifically, I need the app to have voice calling capabilities, and it should be able to handle 6-10 concurrent calls. Simply, this will be a softphone that connects with FreeSWITCH and it is used for inbound and outbound calls control. Skills and experience needed for this project include: - Proficiency in Flutter development - Knowledge of FreeSWITCH - Experience with voice calling capabilities - Ability to work with existing designs and implement them into the app If you have any suggestions or ideas for the app, I am open to hearing them as well.
I use a fusionpbx softswitch and I want help in overriding or replacing the hangup cause from 480 to 503 and I couldn't do it, pls bid only if you're a fusionpbx or a freeswitch expert
Require a experienced VoIP develper to build number purchase portal -Must possess strong knowledge in backend, database and serverside, API Integration -Must know React, NodeJS, API, DB Connection, MySQL, MongoDB and Freeswitch systems. PLEASE IGNORE THIS PROJECT AND DO NOT BID IF YOU ARE NOT FULLY FUNCTIONAL IN ABOVE MENTIONED, Thanks
I am looking for a freelancer who can help me with ASTPP configuration and voip setup. Specifically, I need someone who has experience working with FreeSWITCH, as that is my preferred VoIP platform. The project will involve call routing, so experience with this feature is necessary. I only require assistance with software configuration, and do not need help with hardware setup. Ideal skills for this job include: - Experience with ASTPP configuration and VoIP setup - Proficiency in FreeSWITCH - Strong understanding of call routing - Familiarity with software configuration If you have the necessary skills and experience, please submit a proposal for this project.
We have a FreeSWITCH load issue with media (RTP), and need help resolving it. We are using FreeSWITCH's latest version. Without Media (RTP): 1500 CC (5% CPU Usage) With Media (RTP): 400 CC (150% CPU Usage) We want to achieve 1000 CC with Media (RTP), and it should not take more than 5% CPU. PLEASE BID IF YOU HAVE WORKED ON SUCH ISSUE IN PAST.
I have project c++ using websocketpp library - this a mod of freeswitch - using websocketpp to connect remote client and send data I need guys expert c++ to fix some typo and suggest me clean code of this mini project because im not family with c++ typo. You guy will checkout code & remote to my computer for fixing & running project
Looking for a consultant to work on FusionPBX (FreeSwitch) with Start Trinity SIP Tester so that we can simulate calls (Audio/Video/Callforwarding) scenarios. FusionPBX is the wrapper around Freeswitch with GUI. Anyone with similar expereince and in Telecom will be preferrable. It is very urgent for a project of 5-6 days. Interested Candidate, please reply soon.
As a developer, I am looking for a skilled professional capable of providing development services focusing on FreeSwitch configuration. No existing code will be provided, so the individual chosen should be proficient in working with the platform without any starting points. I am counting on the ideal candidate to bring knowledge and skill in the development space to make this project a success. I need candidate to develop WEBRTC Server with a webrtc client to integrate with different CX platform. Post completion of the project, I will need proper documentation in order to install WebRTC server internally for my project.
Hello, i am looking for someone could help me install call center script. Will be provide full installation tutorial. Installation come with auto file. (One click installed .sh file) Please only take this jobs if you could install immediately. Server is ready. I attached the installation step and file on below...please review
Finishe FreeSWITCH with DialogFlow integration The current status is: - FreeSWITCH installed and configured 100% working - UniMRCP module purchased, installed and working for Google DialogFlow, STT and TTS integration licenses. () The job is: - Finishe the configuration to complete the integration with DialogFlow. It's partially working. Connects, recognizes what is said but does not vocalize the response. Deadline for delivery: Mar/08/2023
Need a custom CDR and DID report page. Please DM me for more info. Please note that a video interview is required before this project begins.
- Develop CTI pop-up in salesforce and upload it on salesforce marketplace - Use or any other method to manage SIP registration, Call, Recording and other necessary features(Outbound call, Inbound Call, Blind Transfer, Attended Transfer, Mute, Unmute). - VoIP PBX is based on FreeSWITCH Language -Javascript, Node.js, APAX (Or any other needed for salesforce CTI pop-up)
We are looking for freelance expert who can work an a specific set of agenda to explain and deliver it's outcomes
Hi, We have OpenSIP (with WSS) with RTPEngine configured but we are not able to make audio calls working for the webrtc based client. Our flow of calls is like this: WebRTC client -> OpenSIPS -> FreeSWITCH The system is deployed on Azure. We are looking for experienced person who has done such work and quickly help us.
What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time (9a.m. ...
Freeswitch amoCRM intgration to enable voice call on the amoCRM
Hi We are trying to find out the possibility of developing a middleware (B) for our system. We have (A) a Voip Switch (Originator) (C) A VoIP provider (Terminator) (B) will be sitting in the center and 'listens' to each Ring Back Tone (RBT) when a call is established and 'ringing'. A--B--C Originator -- Middleware -- Terminator RTB frequecy will be based on standards according to Internation Telecommunication Union (ITU). B will reject calls when RTB frequency are not met.
We would like to setup Signalwire video conferencing and integrate it with our HoduPBX freeswitch multi tenant platform
...a step-by-step guide for configuring FusionPBX/FreeSwitch to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown which will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using FusionPBX/FreeSwitch or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using FreeSwitch with STIR/SHAKEN ()
We need a so-called "banner information" for our own web-based control panel, which is connected to the FusionPBX / Freeswitch telephone system. This means that you can double click on the visually displayed subscribers and then add text there, which is then visible in the banner. In addition, background color and font color should be customizable. Also a link is to be opened by means of right-click and "open link" or so. The text, which one can write into the banner remains to be seen so long on the surface, until the call is separated. These data like URL, color for the background, font color, etc. should be extractable from the description field from the destination management at Fusionpanel. That means that we enter there a hexadecimal code color for the bac...
I am looking for a lua developer who can help me to customise something in my VoIP server. It's lua script which do the functions.. and also use mysql database. Experience in the freeswitch server will be an advantage. Thanks
We are looking for an engineer who can assist with FREEswitch Fusion PBX. We need to update voice messages for 5x PBX users (Christmas greeting messages). Change extension labels for 5 customers and install new extensions and hardware for 2 PBX users. If you can offer other services around FREEswitch Fusion, we would be interested to learn your skills and what you can offer to us.
IT Service im Raum Hessen/Bergstrasse Virtualisierung auf Basis von Proxmox un VMWare,Firewall Konfiguration mit OPN Sense,Windows Server Umgebung, Docker und Kubernetes, VoIP mit Asterisk und Freeswitch
Hello, we are a hosted voip service provider and until now we have been using FreePBX and Asterisk, we are looking at moving away from FreePBX. We are open to either Asterisk or FreeSwitch, whichever meets our requirements listed below: Good day, We are a VoIP Solutions Provider that is currently looking to move away from our FreePBX systems we host for clients to a Class 5 Softswitch/PBX. I have compiled a list of features we want but don’t currently have with FreePBX and then the top features of FreePBX that are an absolute must have in this development. This would need to be a linux based platform capable on running on multiple dedicated servers with iSCSI storage. Ideally, we would like the platform built on a AlmaLinux or Rocky Linux. We would want the ability to ha...
Requiero implementar seguridad a un servidor en astpp, la seguridad a implementar son los escaneos de extensiones sip, escaneo ssh, httpd, y cualquier otra sugerencia que propongan. por favor quien no aya trabajado con esta plataforma que no me haga perder mi tiempo. I require security to implement a server in astpp, the security to implement are the sip extension scans, ssh scan, httpd, and any other suggestion that you propose Please, whoever has not worked with this platform, do not waste my time.
WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk
Hi, We need someone who can upgrade our FreeSWITCH and OpenSIPs to the newest stable versions on Amazon AWS. Currently we use FreeSWITCH version: 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) and the newest stable release is 1.10.8 We also need OpenSIPs upgraded to the newest version 3.3.2 we currently are on: 3.0.2 (x86_64/linux) This is a live production server so it will need to be done pretty quick in a couple hours or so. If we work well together I will have many more ongoing tasks involving FreeSWITCH, OpenSIPs, our PBX and other issues, our main telecom engineer/developer was in Ukraine and we have not heard back form him in months. Thank you! Thank you!
Need to create a system-initiative chatbot (quizbot) using VoiceXML. And also want you to create a user-initiative chatbot (QA system) using the NLTK chat modules (with code that you add to store information or test other conditions, as needed.) Note: you can find VoiceXML documentation online in many places.
Hi Arshad N., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details
Hi Aqs Y., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details
...using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. Our Soft-phones are made with React.js I need a person who knows what they are doing, we also use OpenSIPs so the codecs in OpenSIPs might not be correct but this is just a guess. Can someone solve this for me, I have a hard time getting honest developers here, it seems like everyone says they can fix it, then I waste a week with them and have to cancel and look for a new developers, please only bid if you are truly an ...
Various tasks in freeswitch. Requirements - understanding, communication, desire to progress on the subject. long-term cooperation
We want to provide our cloud switchboard solution via freeswitch. For this, I would like to discuss the project with people who have experience in Freeswitch and API development.
Google Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.
Connect the Web Application with API and Dynamic Data with the Freeswitch / FusionPBX System.
