Flexisip freeswitch jobs
develop a dialplan/RPC that transfers the bleg of a call to the play back of an audio file. and release the a-leg to be hung up. after playback of file to bleg, the bleg is disconnected/hungup
Nehos is an Australian Broadband Internet and Telecommunications Provider offering competitive, reliable and quality services to Business. Our main website - We have been using OpenSIP’s (single instance) as the core of our voice infrastructure for the last 10 years along with Freeswitch gateways but the current setup does not have much redundancy. We have attempted without success to upgrade our current setup to OpenSIPS 3.1 as a cluster for redundancy and load balancing. Our current setup is single public IP address using a destination NAT rule on a Mikrotik Router to route SIP traffic to a single OpenSIPs instance. All servers are running on Proxmox LXC containers. OpenSIPs is using a MariaDB cluster along with RedisDB for storing data. We need to be able to fail
My team installed flexisip on ubuntu18 amazon aws server , installation is completed but we are unable to start flexisip as it shows error unable to locate file in /etc/flexisip/ , another error which we found in logs is some ip and then forbidden ip /forbidden x.x.x.x: forbidden This is fixed project and i don’t know much about these things just message me with a purpose solution and if find you capable only then i will share my server credentials because it is main server of our company and have company data Please do not waste me time i only offer if you have experience in installation of flexisip or linphone
Need to install linphone flexisip in my linux (Ubuntu 18.04). Will have to walk me through the technical issues. Please bid only if you are experienced with package management and linux environment. In server setup we have proxy server not connect showing error:no credentials matched realm or no realm
I am providing sales & support to my customers who come to my website. I need to set up something with which customers can call me directly from my website and i can receive their calls via my browser or admin panel or any customized app or android phone softphone or app. clet me know how you can help configure this. 1) I am looking for a preferably free soft phone solution could be freeswitch or webrtc,zoiper or whatever as far as my basic requirement is met. 2) what will be the infrastructure needed ie. linux machine etc. 3) will they be able to directly call me from my website. Please share any demo or sample snapshots to better understand Kindly explain in as many details if possible or let know if you have any questions
Hi, I am looking for a softswitch class 4/5 based on opensips and freeswitch. Could you create this and what is your price ? Regards
I need a softphone (ish) application that can monitor a Sip extension (freeswitch/fusionpbx) display a popup showing incoming call with caller ID number and name. When the call is answered by clicking the answer option on the popup or when answered via the IP desk phone (yealink t46g) I need it to open a url in a default web browser and have the ability to append the callerID number at the end of the url. (Api integration, and the ability to open a program would be great, but not needed at this time) When the call is answered by clicking the popup I need it to control a yealink phone (t46g) send the call to the yealink phone as an answered call on speaker phone... Click to call on the computer to be able to dial to the yealink phone is also needed. What I am trying to do use to be ...
Hi Amrit M., I interested in updating our mobile app for our freeswitch based system. I'd like to include text messaging as well. Can we talk further?
need to install and configure a system that will allow termination of sms using 50 modems connected to a usb, actually i use linux with chan dongle and astersik fro voice termination, but it does not support sms termination. if it can support it it will be great. otherwise i can use a different module/platform, like freeswitch, Ozekisms, or also Diafaan configuration on windows is acceptable.
Hello, need voip expert , need softswitch with 5000 CC , no reporting , no extra feature . One Customer >> My Switch >> ONe Vendor
We, here on VMAX Digital, want to integrate our Softphone App with Push notification services. And for that, we need to setup a Flexisip push notification server.
Hi Web&Mobile App Developers, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for two developers with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Pur budget is about 500$/month each. Please ping me if interested! Thanks a lot for you time.
Hi Saif, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for a developer with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Please ping me if interested! Thanks a lot for you time.
Hi Arpit, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for a developer with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Please ping me if interested! Thanks a lot for you time.
I am looking for an EXPERT with opensips and astpp with multiple freeswitch to set up a complete switch in a cluster with centralize DB and billing or . When optimize the only limitation of Concurrent calls and cps would be hardware You must use the latest stable versions of everything. you must provide sample of similar work to be considered for this project. Do not waste my time if you have not done this before
I want somebody to install the Flexisip server software from linphone on a machine. Installing Flexisip is not and easy task, if you have not done it before, chances are you are going to struggle at best and at worst you will fail so please save us both the time. The server must function EXACTLY as the demo server on the linphone demo android/ios app. This task should not take a seasoned professional more than an hour to complete. I will supply SSH details to the server, you choose what operating system you want, debian/centos/ubuntu
Need to tune configuration for iternal domain calls .
Hie I need someone to deploy witch on my AWS server.I need someone who can configure an Outbound call center based to my requirement
We need to setup a ITSP phone number to BigBlueButton conference bridge, the purpose of this is users can join webconference meeting from their Mobile phone(call). For this we need to configure FreeSWITCH to receive incoming calls via session initiation protocol (SIP) from our nexmo(ITSP) provider. We are looking for someone who has deep knowledge in freeSwitch and BigBlueButton.
Hi Fellow VoIP experts, We are working on a leading web-conferencing open source platform for online learning. This platform uses Freeswitch with Opus codec through WebRTC; so directly through the browser. We made some tests and the audio quality is a lot better on Jitsi, we want to achieve the same quality through our platform. If you are a certified VoIP engineer or have a lot of experience configuring Freeswitch for web usage with opus codec through WebRTC we would be glad if you could help us. So your mission is to help us understand Freeswitch mod_conf opus codec and achieve a better audio quality result.
Minor mod_xml_curl to fetch dialplan and gateway config for FreeSWITCH servers. Implement Fail2Ban and UFW. Have the APIs in place. Just need some help on how to fetch them and implement them in FS.
Hi Seby Francis K., I noticed your profile and we would like to have some help in freeswitch configuration. Would you be interested?
Freeswitch installation need to be configured for round robin distribution
I need help on building a simple API for FreeSwitch. It is a small and quick project I want to implement at least 4 endpoints reloadacl reloadxml status sofia profile status A server with loopback4, node and freeswitch will be provided.
Hi Aleksei B., have some flexisip work for you. Also linphone customization but i think that is less your field. let me know Thx
Hi there I have a Softswitch name ASTPP V3.6 which is running on freeswitch There is failover gateway options which I try to use it but its not working want solve that. And also one more issue that sometime SIP account in unregistered automatically. Thank you
i need filter on asterisk about did and clid number. We are using this project on maniplation temination voip traffic
Hello, make push gateway for flexisip to connect to sipwise c5 PRO
Hello, We are looking for a developer who really knows ASTPP and FreeSWITCH, we are wanting to modify ASTPP and add a lot of API's to it. Must be an experienced PHP and telecom developer. This is for long term ongoing work. Thank you!
We have our mobile app built with linphone sdk and we want to use flexisip installation on our computer. The goal is that you will set it up to the point that we can use it in our mobile app to work properly
I need someone to install and configure flexisip and all required components to work with linphone.
Hi Rajib M., I hope all is well. I have a similar project to the one you made before "Linphone Flexisip and VoipPush Implementation for iOS" ://www.freelancer.com/projects/mobile-phone/linphone-flexisip-voippush/&sa=D&source=hangouts&ust=1597997370615000&usg=AFQjCNGXV7VRhI5CurNSMtZ11tncWP9YfA Are you available and interested? regards/ Adel
Hello, We are a very small telecom startup and need an experienced developer to set up and install ASTPP, FreeSWITCH & OpenSIps and set them all up together, link the DB's and add our telecom carriers and get everything working 100%. If you do good work we will consider hiring you for ongoing work to maintain, run and modify it for us. This all has to be installed on AWS so you must also have knowledge of AWS and must have excellent knowledge of ASTPP, FreeSWITCH & OpenSIPs and Voip Telecom. Thank you!
Linphone for iOS and Android is up and running (able to to receive push, background calls and SMS) Our developers can provide some support for Mobile Applications a...receive push, background calls and SMS) Our developers can provide some support for Mobile Applications and asterisk, they have limited knowledge related to push notification. - Flexisip must be installed and integrated and all modules must be in working - Flexisip must be able to communicate with Asterisk to send or receive SMS (custom work) - Flexisip must be able to send push to iOS, Android Devices Please do not bid if you don't have good understanding of the project - Flexisip Overview - Flexisip Wiki Wiki
Hello, We are a small telecom startup looking to hire a long term developer for ASTPP, Freeswitch & opensips. Must know PHP, Codeigniter, MySQL. to Start it will be around 20 hours a week but if your really good we can move you up to more hours per week. Please send your reume with experience.
When we receive a fax, we see the call in the Active Calls page, and then it disconnects automatically within a second. A year ago our fax server was working perfectly with the same configurations. We didn't use it for a little while but now when we are trying to, it is not working. We need someone to quickly debug this issue and fix it. we can share with you the Fax log during the interview and share it with you after you are awarded this job.
We want Freeswitch service of Bigblueserver to be set on Saparate Freewitch server
LUA script I'm looking for a IVR LUA script : Step 1: Play Sound (What is your employee number Step 2 : See in Mysql all employee number (active) Step 3 : Load the employee in the IVR Step 4 : The call enter the employee number Step 5 : The script verify in Mysql if the employee are In or Out Step 6 : The system put the employee out if are in and put in if are out in the MySQL Kind Regard
we want to separate Freeswitch server on our bigblue button sever
I need someone to install and configure flexisip and all required components to work with linphone.
Hi Friend, We have stuck at our PBX to receive more call, the problem is description here: My Server is : 12 VPU, 14G RAM, almost free when error occur We try so many way to improve the PBX but not suceess I need an export support me config to receive more call from PBX The info of system: - FusionPBX - FreeSWITCH version: 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) - PostgresQL Database
I have a voip project need use the LinphoneSDK + Flexisip server. I have setup the Flexisip server but I found if we use account. Everything works smoothly. call from LTE to Wifi. Works fine. But switch to the sip server we created. It only work with in the same local network. No matter we enable the ice/stun or not. I think there some configuration thing I miss. also there one more thing about the mysql database authentication.I try to enable the mysql database authentication. But it not work. The file storeage authentication works fine. I need some developer with the experience about the LinphoneSDK + Flexisip server to help me. Thanks Frank
I would like to configure flexisip sever with push notifications along with asterisk and any softswitch. Prospect freelancers should have proven experience with SIP and Asterisk and ability to complete job on time.
Hi Gaurav S I searched on google regarding flexisip configuration and i found your profile. I would like to talk more about it. Ping me manish.v@
I’m Looking for developer for a VoIP platform with Billing solution. To save time, I’m willing to use Opensource FREESWITCH and ASTPP as core of the system. The customizations and integrations should be coded with PHP/MySQL, and being easy to integrate with a Wordpress site too. Not a CUT & PASTE job. I need a developer who fully understands the programming languages used, experience with Softswitch platforms, and the final job has to be FULLY TESTED, Documented on specific areas (inventory list of files, scripts, plugins) that were modified, customized, added, or linked to the Open Source code so they can be properly managed and updated in futures upgrades of Freeswith/ASTPP. For now platform will be used in just one Server, but it must support redundancy and...
