Freeswitch kamailiodela

We are 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 ...

I am looking for an experienced Kamailio administrator to help configure my Kamailio server according to my desired specifications. The current setup of my server is highly customized and requires general configuration assistance. Through this project, I am looking for someone to fine-tune the settings, features and modules to ensure that it operates with optimal efficiency. Communication between us will be done primarily via instant messaging tools such as Slack or Skype. I'm comfortable with either option. I'm open to hearing any additional suggestions or recommendations you may have that could improve the setup of my server. With your experience and expertise, my Kamailio server is sure to be in peak operating condition.

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

...configurer Kamailio sur mon serveur et orchestrer plusieurs serveurs Asterisk. Mon objectif est de recevoir des appels de Twilio et Telnyx, et de mettre en place des règles entrantes et sortantes pour gérer ces appels. J'ai déjà installé un docker de Kamailio sur mon serveur, mais je ne sais pas comment configurer les premiers réglages. J'aimerais utiliser le mode file plutôt que la base de données, ce qui signifie que j'aurai un fichier de réglages contenant les adresses IP de mes serveurs, les adresses de Twilio et Telnyx, ainsi que les règles entrantes et sortantes pour gérer les appels. Je suis à la recherche d'un freelancer expérimenté en Kamailio...

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.

Installation of A2Billing latest and asterisk on a debian server. For future integration with linphone SDK I want someone who has past experience in configuring asterisk to work with A2Billing. Your job will be to and install A2Billing billing 2. install and configure Asterisk with cdr to be working on A2Billing I should be able to an account from A2Billing and register on a softphone. 2. Make calls using that account 3. CDR should be made in database table and I should be able to see it in A2Billing billing. You also need to tell me which configuration files were changed so that I can do this myself next time in case of server failure or new installation.

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

I'm lookin for of a skilled engineer to manage and deploy a SIP & Kamailio on docker compose that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engineer is experienced with the SIP & Kamailio & docker and be comfortable dealing with any issues that may arise. The engineer must also have experience in deployment on docker & docker-compose and later on administration of this system. This position will require some extra work every week, but I'm ready to discuss the amount of work needed with the chosen expert in order to get the best possible results. We usually discuss about our requirements, w...

We are looking for experienced professionals to help us build a VoIP SBC system, using Kamailio/OpenSIPs preferred. The services we would like to set up include hosted PBX, call termination, and call routing. We do have an existing infrastructure which needs to be connected to, so knowledge of this process is needed. We are also interested in advanced features such as call quality monitoring, packet capture, security measurements etc, analytics etc. There is a billing system which is a sort of SBC itself, but very small and only intended to bill call termination. Also, we have a bunch of Asterisk servers spread all over the internet acting as PBXs and sort of B2BUA proxies/gateways for the rest of infrastructure. The idea of this project is to build a geo-redundant system having...

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.

I'm in need of a skilled engineer to manage and deploy a SIP & Kamailio & Asterisk solution on-premise that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engineer is experienced with the SIP & Kamailio & Asterisk solution and be comfortable dealing with any issues that may arise. The engineer must also have experience in deployment on docker & docker-compose and later on administration of this system. This position will require some extra work every week, but I'm ready to discuss the amount of work needed with the chosen expert in order to get the best possible results. We usually di...

I am looking for an experienced freelancer to help configure my existing Kazoo Kamailio 4.3 to work with the latest version of Kamailio (5.6/5.7). I already have Kamailio 5.7 installed and running, but have some issues getting Kazoo Kamailio 4.3 to work with it. I need assistance specifically with the configuration of Kamailio for this project. More tasks will be available for long term cooperation for the right candidate.

...X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 3) ALL TCP SIP traffic on 5062 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC) 4) Listen for incoming SIP TLS traffic on 5089 on the LAN IP X.X.X.X. and proxy this SIP TLS Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 5) ALL TCP SIP traffic on 5089 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC) 6) The other phones and the sip trunk providers still use UDP 5060 and should not be proxied via Kamailio , and instead should communicate with the Asterisk server directly. Deliverables will provide the complete or to allow the remote phone to register

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

The standard Kamailio Proxy-CSCF IPSec module 'ims_ipsec_pcscf' binds to n ports(defined in config) when listening for an IPSec connection. Each connection is on its own unique port, both client and server side. I need the following functionality: The Kamailio P-CSCF Server needs to bind to one port only, and keep track of each connection via client ip address (or any other connection-specific identifier, whatever is easier to implement). I need be able to then access that connection's meta variables via config. Eg: Client 1:5991 -> Server 1:59000 Client 2: 5998 -> Server 1: 59000 This should be relatively easy to implement for any adept C developer with networking knowledge. The module to be modified is attached to this request.

...pipeline support, we take pride in our ability to take on a wide assortment of complex cloud-based projects and turn tech challenges into positive outcomes. Some of your day-to-day responsibilities will be: - Produce high-quality code in VoIP backroom environment, desktop telephony solution, and WebRTC telephony implementation using C, C++ - Implement new features and maintenance in Asterisk, Kamailio, and in-house telephony libraries. - Collaborate with peer software developers to design, develop, and implement industry-leading Web-based call-taking application. - Participate in all phases of the software development cycle as part of a multi-functional Scrum team. - Participate in the architecture of a scalable and robust software deployment model. - Implement and mai...

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.

Required administration training on Freeswitch

Hi; I am a private IT enthusiast, who is interested in learning more about IT. I have recently come aware about Asterisk, Kamailio and Opensips. I would like to build a lab using virtual machines to see, how all of this works. What I am looking for is a kind of tutor, who could guide me through the steps to build this lab and get it working, while explaining to me tha essential things that I need to know. I am not a professional, so I do not need much details, even though sometimes I would have to get some. If anyone is interested in helping me building this lab once with Kamailio/Issabel and Kamailio/FreePBX and also using Opensips - connect it to Jitsi or MS Teams, do not hesitate to contact me and tell me, how much would it cost. Thank you.

- 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)

Sip Refer To from remote server to asterisk tought kamailio Tengo problemas para manejar un Refer To remoto desde un proveedor hasta asterisk. La respuesta del servidor siempre es 481 Call / Transaction Does ...... .

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.

Hi everyone, I'm looking for experienced VoIP Engineer to give us helping hand on real-time speech recognition system we are building. Our stack: Debian,...for experienced VoIP Engineer to give us helping hand on real-time speech recognition system we are building. Our stack: Debian, Asterisk v.16-19, Kaldi + Vosk, Python/Javascript (node.js) The main problem is that we are unable to pick a real-time stream of RTP which is a plain UDP, transform it to Websocket data and send to the Kaldi server to recognize. We would prefer to get it done without Kamailio and RTPengine for now, just plain Asterisk possibilities like UnicastRTP, ARI etc. So we are seeking for experienced Asterisk engineer who can give us a valuable hint, share experience and/or write some code for us. Thank ...

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

...goal of this project is to create a step-by-step guide for configuring Kamailio (a free and open-source SIP server) to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown as it 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 Kamailio or Martini Security's offerings. 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 Kamailio with STIR/SHAKEN. Martini Security offers a certificate enrollment client

...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.

I have a white-labeled Linphone application for both iOS and Android completed and use Asterisk servers. Push...Android completed and use Asterisk servers. Push notification for incoming calls is causing me challenges. I'm looking for someone to hire to: Walk me through setting up my Apple developer account and Firebase to send push notifications to my Linphone build. Walk me through associated modifications to Linphone build (integrate Google plist, etc, whatever needs doing). Configure kamailio (preferred) or flexisip (acceptable) to proxy between my Asterisk systems and Linphone on client mobile devices to handle push. We'll use a fresh Debian 11 install that I will provide you credentials for. If this is something that you can take on, please ...

WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk

We installed goautodial v4.0 from iso Kamailio running HTTPD OK SSL certificate OK RTPENGINE Ok Our main issue is the following: 1) Agent need to press (Login to dialer 2 times) 2) Can't register GoIP gateway (SIP Trunk) 3) Can't hear any voice. Only Goautodial V4.0 specialist is required...!!

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!

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