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    1,525 freeswitch jobs found, pricing in USD

    I have instslled ASTPP which comes with freeswitch I need someone to configure WebRTC clients to connect web phones For example And provide " how to " guide

    $36 / hr (Avg Bid)
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    10 bids

    We are looking for an expert who can help us to make our webrtc client working with opensips. We have Opensips as SBC and FreeSWITCH to handle media + call routing logic. If we connect our webrtc client with FreeSWITCH directly then webrtc working well but when we connect the webrtc client with opensips then outbound and inbound calls are not working. Please bid only if you have worked on similar task.

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    ICTCore core is open source freeswitch based unified communications framework for developers and integrators to rapidly develop ICT based applications using their existing development skills Following you will find more details about ICTCore communciations framwork the Fax server software developed over ICTCore communications framework

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    VoIP Expert, VoIP developer for Freeswitch, Fusion PBX, to develop an API that allows integration in our application, where users should be able send and receive text messages, initiate, answer, transfer or put on hold a voice call, listen to VM's and call recordings, be able to send and receive Fax.

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    We need a person who works on freeswitch lua and asterisks who can work with us to maintain astpp server

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    We want to set up a multi-tenant cloud exchange, and call center structure based on Fusionpbx or directly on Freeswitch. We are currently doing the same work on Asteriks. We are interested in Freeswitch based builds due to management difficulty. In order to integrate the structure we have already used into my new site, we need the following developments. Our wishes. 1. When a new customer is created on our own CRM application, it will create requirements such as Domains, Gateways, Inbound Routes, Outbound Routes, Extensions on Freeswitch via API. 2. In addition, Active Calls, and Extensions states that I have mentioned below are given to us via WebRTC API. DIALING RINGING EARLY ACTIVE HELD RING_WAIT HANGUP UNHELD NULLInternal states( Busy call some info) 3. Providing A...

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    Hi it's a freeswitch on Debian 10 fresh installation

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    This requires setting up a call center with FusionPBX/FreeSwitch with the following points considered - a queue for incoming external calls (I'll update the destination_number directly in the Dialplan); - each agent can receive only 1 concurrent call from the queue; - if there's no available agent, the caller listen to a recording until there's an available agent; - the agents must be called in a defined order (first agent 1, if available, if not, agent 2, if not, agent 3, and so on); - one extension to listen to any agent's call in real-time; - recording of all the calls; - shortcut to transfer an ongoing call to another user; - shortcut to pause/resume receiving calls from the queue (but not from another user);

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    I need someone to configure a Freeswitch/FusionPBX server. It needs: - a queue for incoming external calls (I'll update the destination_number directly in the Dialplan); - each agent can receive only 1 concurrent call from the queue; - if there's no available agent, the caller listen to a recording until there's an available agent; - the agents must be called in a defined order (first agent 1, if available, if not, agent 2, if not, agent 3, and so on); - one extension to listen to any agent's call in real-time; - recording of all the calls; - shortcut to transfer an ongoing call to another user; - shortcut to pause/resume receiving calls from the queue (but not from another user);

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    I'm looking for someone that can help me troubleshoot call being blocked on freeswitch and fail2ban . Ineed someone that is really good at troubleshooting acl list etc.

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    ...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

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    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...

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    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.

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    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.

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    Hi David T., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    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.

    $10 - $40 / hr
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    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 )

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    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.

    $15 - $25 / hr
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    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

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    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...

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    Looking for someone to support FusionPBX / FreeSwitch must have experience with PBX especially FusionPBX.

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    Need to configure the Free Switch for the SMS broadcasting

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    ...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...

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    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...

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    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

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    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.

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    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.

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    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

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    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.

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    Radiusd didnt bill a call after a hangup 18 while call was active and did 900 sec only experts

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    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

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    My requirement is to integrate Web-Dialer to Freeswitch to make the calls easily. By default Freeswitch has a facility to make calls by the backend or we can configure SIP details to third party dialers in making a call. For an easy way I need this Web-Dialer to be integrated to Freeswitch. Does any one help or suggest in this.

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    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

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    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...

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    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...

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    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.

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    Salam Ibrahim, I'm looking for a FreeSwitch/WebRTC expert.

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    Am looking to create a WHMCS provisioning module for FusionPBX using API. Please provide example of WHMCS modules you have provided

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    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.

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    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...

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    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

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    Looking for someone to assist with FusionPBX/FreeSwitch

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    Hello, Rajesh Kumar S.. I noticed your resume and want you to participate in my project. We can discuss the details of the project.

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    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.

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    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.

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    VoIP kazoo /Freeswitch LB / vs Kamailio setup (custom API) page to modify SIP invite programming

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    We want to enable ZRTP end-to-end encryption on a Freeswitch SIP server

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    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.

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    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

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    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...

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