Freeswitch openbts jobs
...Registration servers Freeswitch as the Media Server 1. Instal Kamailio OS: Debian 11.3 Question: Does MariaDB required? I prefer postgresql 2. Install Freeswitch from master( I will do it) 3. User Registraion from postgresql database, I will provide the Database and table Eg: Customer 1 - user1 at , user2 at Eg: Customer 2 - user1 at , user2 at Eg: Customer 3 - user1 at , user2 at DNS Srv Records - Cloudflare and I will configure. 4. Customer1 can call their users only 5. Voice Mail Sent to Freeswitch Customers - Use Own Music On Hold Calls and IVR - Tenant I will provide the debian server, with freeswitch installed. Setup the
Night Shift IST Product: CallHippo. URL : CallHippo was launched in 2017. It is an intelligent VoIP (voice over Internet protocol) service provider for busi...create thought leaders in the business ecosystem Exp : 1 yr + Job Description - Responsible for SIP Development experience. -Involvement in SIP and webrtc integration. - Responsible for VOIP software development. -To work on Queue, IVR and Voicemail related applications. -Responsibility of Freeswitch installation, configuration and troubleshooting. -Deployment of multiple instances of Freeswitch using a load balancer. Requirement -2+ years of experience in FreeSWITCH or other related VoIP technologies. -Good Knowledge in PBX, SIP, RTP protocols. -Experience with VOIP Software developme...
Hello, we need to setup an ICT fax server. Centos7 is installed, along with ICT FAX and freeswitch. Need someone to configure it and successfully finish the setup.
I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want to move our SBC's from FS to OpenSIPS and am wanting to do the work on our own. We will require -- assistance with creating rules in OpenSIPS Control Panel -- assistance with loading the proper modules -- training on how to write the proper logic for routing calls We do currently have 2 running systems with OpenSIPS that is maintained by a 3rd party which has been doing great for us. All updates need to be understood that the current config must still work after completion.
Hi David T., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want to move our SBC's from FS to OpenSIPS and am wanting to do the work on our own. We will require -- assistance with creating rules in OpenSIPS Control Panel -- assistance with loading the proper modules -- training on how to write the proper logic for routing calls We do currently have 2 running systems with OpenSIPS that is maintained by a 3rd party which has been doing great for us. All updates need to be understood that the current config must still work after completion.
what I need to edit to fix that , my freeswitch is sending ack to supplier if no reply in 60 sec call gets disconnected how do Ignore that or make it 3 min instead of 60 sec/ or how to make asterisk make fake 200 when ack is requested ( When your stupid supplier is not replying to ack )
I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want to move our SBC's from FS to OpenSIPS and am wanting to do the work on our own. We will require -- assistance with creating rules in OpenSIPS Control Panel -- assistance with loading the proper modules -- training on how to write the proper logic for routing calls We do currently have 2 running systems with OpenSIPS that is maintained by a 3rd party which has been doing great for us. All updates need to be understood that the current config must still work after completion.
We already have a Voipswitch developed on Astericks and also FreeSwitch and we also have IOS and Android apps that delivers Voip calling from any country. We experience Voip blockage in some countries eg UAE and Brazil etc. We need you to be able to build or create a tunnel or add mediator server that will act like a proxy that will bypass the voip blockages put by those countries . Thereby allowing voip calls to flow to and from our softswitch without loosing call quality or volume. The solution must also mutate so that if the country notices our service and blocks our app or IP. Your solution will mutate or change IP so that customers will always be able to make his or her international calls without blockages
I have three requirements that I would like to you see if you can quote me for All are based on Fusionpbx/Freeswitch. Currently we are using Fusionpbx 4.4. 1. Develop a sticky agent feature in the call center module. requirement would be Lets say a Caller has called to the system and the call was answered by Agent A and assuming the caller called within a configurable time internal (lets say 24hours) and if the Agent A is still available to take the call on that particular Queue the Caller came in , the Call needs to be routed to that agent only (Priority given to that Agent). b. If the agent is not available or busy at that moment then the call can be routed to the strategy as selected on the Call Centre Queue (Random, Idle agent etc). Please do let me know on the commercial a...
Looking for someone to support FusionPBX / FreeSwitch must have experience with PBX especially FusionPBX.
Set up 4g LTE gsm base station using blade RF and srslite or openlte. And 2g station using openBTS
Need to configure the Free Switch for the SMS broadcasting
...experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch Skills: FreeSwitch, VoIP See more: open source technologies pvt. ltd, open source technologies , ringless voicemail, professional voicemail recording, audio voicemail greetings, british voicemail recording, recorded voicemail greetings british, voicemail voice, voicemail email elastix, openser voicemail asterisk, voicemai...
I'm looking for a technical person who has good telecom experience and can help us develop ringless voicemail features. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : [url removed, login to view] You can read about it and tell...ringless voicemail features. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : [url removed, login to view] You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc. Please bid only if you know about and are able to help. Thanks Desired Skills Freeswitch Sk...
Looking to create a High Availability setup for FusionPBX/FreeSwitch. Require the setup of two replicated Postgresql servers, file sync between the two PBX Servers, and setup of pacemaker. Bidder must have experience within setting up High Availability servers. I will install Centos/AlmaLinux on all servers ready for the setup
Hello, We need to implement CGrates with our current SIP infrastructure. The infrastructure in mind is an SBC server connected to a cluster of 2 Freeswitch. We would like to implement this as soon as possible. Thank you.
We are wanting a bespoke billing solution with Freeswitch. Possibly nibblebill or vbilling would be a good start. This solution will ultimately be a class 4/5 switch, be capable of managing DID numbers, multilevel admin/reseller login, and manage billing. Full specs will be detailed at a later stage.
ENG A plugin for VoIP Client is required. We already have the VOip Server (Freeswitch + FusionPBX). Our application is built upon @ionic/angular 5.6.3, capacitor 2.5.0 Our application needs to be deployed in iOS 14 SPA Se solicita crear plugin para proyecto en Ionic Framework (@ionic/angular 5.6.3, capacitor 2.5.0), la cual permita realizar llamadas (solo audio) por protocolo SIP, conectado a un FusionPBX. Su uso debe ser de forma similar al plugin de cordova (cordova-plugin-sip). El plugin debe funcionar en versiones android 11 y superior, ios 14 y superior
We currently use Twilio/OPENVBX but are searching for an upgrade as the system is 5-6 years old and has some bugs. We are a small office, but would like a system we can scale. Asterisk, VoiP, Freeswitch, etc.
Radiusd didnt bill a call after a hangup 18 while call was active and did 900 sec only experts
Hello we are using a open source fax portal that is based on freeswitch technology. Faxes are coming to the mailbox phone number at the server side and we can manually download it BUT its not going to the associated email address. Seems like some script issue. We used to get it at our email address. Who can help. Must have some experience. Its a simple task for the qualified person. We may become your customer for our other similar needs. I think it uses php and angular for UI
I'm starting to use Accredible to generate the certificates for VoIP School. I want to create custom certificates and badges for the school. The current certificate and badge (I think they are really ugly) i did it myself. The badges will have the title Verified (Asterisk/FreeSwitch/SIP) professional I'm uploading the current certificate and one badge I did (as I said terrible) and the VoIP School Logo
We need a WebRTC client SDK that can be implemented in 3rd party projects. Needed functionalities: WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAG...WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/disable video during a session (during an ongoing call session re-INVITE and disable or enable video) WebRTC signaling plane: - SIP over WebSecureSocket (will connect to a sip server as Kamailio/Opensips/FreeSWITCH) WebRTC media plane - codecs: 711, opus, VP8, VP9, H.264 - DTLS/ICE/SRTP API facing side: - provide an easy and comprehensive API for quick integration into 3rd party...
We are looking for an experience company or a team capable of development and future support of a custom Kazoo PBX. Experience Requirement: -Must have built something similar and demo the solution with Kazoo -Previous development with Kazoo, Freeswitch, Kamailio -Examples of previous development of telephony systems based on these platforms -Excellent understanding of pbx telephony, dial-plans, security -Excellent understanding of hosting technologies -Excellent understanding of Database (SQL) with High Availablity (HA) & data replication/sharing, DRDB IF YOU HAVE NOT DESIGNED AND DEVELOPED A SIMILAR SOLUTION, do not apply. we require an experienced team capable of full delivery on time and within budget. The initial scope of the project will include creating the following o...
I have found some code ( or ) on Github for a web UI for OpenBTS which I installed on my laptop via an ISO ( from ). It is supposed to be simple but I have the slightest idea where to put the code. I need someone to install the code on the same ISO from which I can reinstall with the new UI.
Looking for a Prepaid real time billing developer for freeswitch the system should not be based on mod_nibllebill or CGrates it should be developed from the scratch.
I have found some code () on Github for a web UI for a program on my server/laptop called OpenBTS. it is supposed to be simple but I have the slightest idea where to put the code. I need someone to install the code or show me where and how to.
Salam Ibrahim, I'm looking for a FreeSwitch/WebRTC expert.
Am looking to create a WHMCS provisioning module for FusionPBX using API. Please provide example of WHMCS modules you have provided
Hi We are looking for someone to Build an Auto Dailer CRM using Asterisk , Freeswitch and NodeJS. Designs and wireframe shall be provided by us along with HTML and CSS of all screens.
I need a developer who can create a WebRTC server using Janus or FreeSWITCH. We have a system that works in two ways: 1(attached graphic). Calls will come in through our DID. Each call is authenticated through our API (handled by a separate developer) and then each user is placed into separate channels after API authentication. The audio stream from the caller will be listened to and processed by our audio tools. The calling system is setup through Twilio SIP trunking. 2. When each user is in a webrtc channel, allow for the ability to have one or two users join via webrtc clients such as our mobile app, browser, etc. (another developer will handle client development). We are NOT building a conferencing app, but we need the ability to accept calls and then later perhaps have a c...
I installed the Astpp Billing but the issue is the inbound calls are failing giving me an error WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'default' for
Hello, Rajesh Kumar S.. I noticed your resume and want you to participate in my project. We can discuss the details of the project.
We need a FreeSwitch expert who can work on the modules, installation, configurations. Note - Want to implement voice changer on the live calls. Only bid if you have experience with the same.
We need a FreeSwitch expert who can work on the modules, installation, configurations. Note - Want to implement voice changer on the live calls. Only bid if you have experience with the same.
VoIP kazoo /Freeswitch LB / vs Kamailio setup (custom API) page to modify SIP invite programming
We want to enable ZRTP end-to-end encryption on a Freeswitch SIP server
Knowledge of the complex issues of FusionPBX does not exist and requires someone with expertise to review, recommend and implement system changes. It will be an ongoing project/T&M position with the right person.
Hi Nsikak A., I saw you already have a click2call developed for a client, my platform is ASTPP 5.0 with freeswitch. I need the customer to have the option to choose to receive the call on the extension or pstn, and after answering the freeswitch sends the call to the site visitor. Whenever both incoming and outgoing calls are PSTN, they are billed in astpp. It can be in the two scenarios you already have: 1. Click to dial by users 2. Click to call Thanks
We are a telecom, marketing and analytics startup looking for a PHP/Laravel full-stack web developer with experience in VoIP/SIP & Twilio or similar Telecom company and having knowledge of telecom and how telecom systems work especially FreeSWITCH, OpenSIPS & PBX’s. You must have the following skills to qualify for the job: PHP Laravel Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge of VoIP, SMS /MMS, Call center style call queues, call conference rooms FreeSWITCH OpenSIPs AWS React We will offer long term work every but only if you prove your skills in the first few weeks. We are only looking for serious developers and to prove that please fill the attached questionnaire document, upload it anywhere and add th...
Looking for someone to setup kamailio / OpenSIPs for the following - Registration pass-through SBC for Freeswitch Servers and Remote Phones... kamailio / OpenSIPs will be hosted inside Amazon EC2 - SBC Setup for phone calls with Least Cost Routing... Freeswitch will use kamailio / OpenSIPs as the gateway and kamailio / OpenSIPs will pass the call to the appropriate ITSP - Setup kamailio / OpenSIPs with local registrations for SIP Clients to handle Video from Door Phones for Remote Clients
intento cambiar la credencial de Event Socket para asi crer un Freeswitch seguro en astpp, pero al parecer no encuentra la direccion de freeswitch, e usado la documentacion de astpp pero aun asi no logro dar con esto esta es la url vim /usr/local/freeswitch/conf/autoload_configs/
I'm looking for a developer who has a lot of experience with FreeSwitch for a long-term project to build it in our cloud environment for many different use cases, therefore it has to be general to begin with and then we will build out for specific use cases. I'm looking for someone who can respond quickly, is flexible and agile with deep experience because the project is going to be asking a lot from a development pov. Time is against us and you will be building in Oracle cloud.
Hello, we are looking for a developer that can handle a couple of projects involving Astpp, FusionPBX, and Freeswitch. If you have experience with Freeswitch and Lua, you should be able to handle the projects. More will be discussed once I get an idea of your experience and hourly rate
I am researching community cellular networks and need an installation of OpenBTS gsm system that works with software defined radio. Should work in India. Skills Required Telecommunications Engineer, C Programming, Computer Security, Embedded Software, Engineering, Linux, MySQL, PHP, Software Architecture, Software Testing, Web Security, Python Script, MATLAB, ASTERIX Hardware, Raspberry Pi
Hi, I have an issue with Call Deflection when a mobile calls into a DDI which is then forwarded back onto another mobile. This is getting rejected when Call Deflection is turned on. Works when off but I need it on to show the A party calling number. Need some to trace and fix my SIP Trunk parameters / settings. Using Freeswitch with the same SIP Trunk it works so it is supported by the carrier. I just don't know the 3CX system enough to debug and fix.
I am looking for someone to quickly put together a recording demo using either Asterisk or Freeswitch where the user is able to do the following: 1. Login into a basic site (will provide access to a DO server) 2. Have the option to start/stop a recording (simple button should be present for the user to click) 3. Once the recording is stopped it should be saved. 4. Ability to view and playback recordings with timestamps. We do not require any options to delete or edit recordings. 5. The user will login into the site on their PC and using a USB microphone will conduct the recording.
We are currently using a custom PBX solution. The PBX server is hosted in the Vultr Cloud and is based on Freeswitch with the frontend as FusionPBX. There are few issues which have come up which needs to be fixed urgently. The SIP trunk is being provided by Voxbeam 1) All incoming calls should ring the SIP softphones 2) Outgoing is not working properly for some regions. Need to diagnose that and resolve 3) If possible, need to configure an alternative SIP trunk in case Voxbeam is failing. The process should be automatic.
