Freeswitch kamailiopekerjaan
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.
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
...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.
Hi I need a hand with freeswitch: I have a client that needs to send calls to my freeswitch but his switch doesn't have username+password authentication. He is asking me to have my freeswitch accept the calls based on his public ip address. Let me know if you are able to help but only bid if you have extensive experience with freeswitch as this is a security concern. Max 100 euros. Thank you.
... • Candidate should be familiar and comfortable with Freeswitch. • 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. • Expert in Freeswitch installation, configuration and... • Competent enough to setup daily call limit and concurrent calls Requirements · Software Development experience in Freeswitch, FusionPBX, Opensips, SIP, VOIP, SDP, TDM, IMS, PSTN, Python, Perl, Linux, and Open Source Technologies. · Strong Technical, Logical and Debugging skills with innovative and result-oriented approach ·working experience in Python, Shell, Perl, Asterisk...
Hello Amrit, We are an Italian internet provider. At the moment we are using a custom Freeswitch cloud pbx calle Hodusoft. We need to customize Linphone to work with. Registration is OK but blf doesn't work and we would some feature like an easy provision and contact. Can we discuss about it? M
A VoIP telephony company is using a kamailio server and a partner company to route incoming calls of the customers. 2 new partner IP's needs to be added in kamailio, which is already setup and is in production, to route the incoming calls. The purpose of the project is to assist the company trainee to verify the kamailio setup and how the IP's are configured (provide the commands, etc.). Then to detail the procedure on how to add new IP's in kamailio (provide appropriate steps and commands). Make sure no service will stop or break while adding the new IP addresses. Assist to restore the services inside kamailio, in case of issue on kamailio server. Verifications, details on configuration, and whatever is needed for freelancer, to u...
We are looking for someone who can fix certain compatibility issues with of Flutter BigBlueButton app with latest BBB server. In the long run we'll need to include additional features as well. Flutter Code: Make sure you know what FreeSwitch, WebRTC means. Without depth knowledge on them you'd get lost
