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    1,280 freeswitch jobs found, pricing in NZD
    Freeswitch 1 day left

    Configure fusionpbx freeswitch

    $34 / hr (Avg Bid)
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    We are looking for a VoIP systems engineer to help complete the development and testing of an internally developed web based calling application linked to Asterisk/Free PBX. - Experience with Asterisk and/or FreePBX - Experience with Apache, CentOS, DNS, hosted services, MySQL - Network design – Working with Firewalls, DNS, Load Balancers - Experience in software as s...

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    Hi, Its installed on Ubuntu . My freeswitch service is not working. please contact if you are expert in it. I already try 1) systemctl start freeswitch 2) service freeswitch start /restart/stop [Removed by Admin] thank you root@freeswitch2:~# fs_cli [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] Usage: /usr/local/freeswitch/bin/fs_cli [-H ] [-P ] [-...

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    We are need for a Fusion PBX/ Freeswitch Admin that can install & test the following services. Multi tenant, Presence/Chat, Real-time Users, Real-time Dial plan, call recording. We would like to hire somone to work by the hour to complete one project at a time. We will create the project, you will quote time, you will complete and document install process. We will start by creating an chat app...

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    we are looking for cluster freeswitch server with load balancer with given below features. A- API Documentation B- Deployment Diagram if available C- System Installation & Configuration guide 1- API Server 2- Freeswitch D- Complete Source code E- Funcationality 1- Live calls 2- amd beep detection 3- Play messages on press 1, press 2, press 3 etc... 4- after press 1, 2 send call to sip extens...

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    I need to customize some litle aspects of the GUI of the ASTPP Billing. This is the customization that I need: 1 - Show active calls on clients login 2 - Show negative "balance" for post-paid clients, exactly like the PRE-PAID clients, not the "credit limit" information. 3 - Create a random callerID for specific sip devices, like that: If I call to the number 55 48 996843116,...

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    Our project is to build web GUI management for FreeSwitch Telephony Engine using python for dynamic Configuration, Dial Plan, user Directory also using PostgreSQL DB

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    we are looking for a developer who can work on cgrates+freeswitch. we want deploy cgrates billing with freeswitch and initiate the call using API and freeswitch auto load balancer

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    Hello i am looking for an expert user on freeswitch pbx system. i need to learn lot of things on this freeswitch depend on my time time need. at the moment i am using fusionpbx for the multitanent. but i need to learn few things like inbound route outbound route dialplan how works etc etc etc. i am a advance user so i am looking for expert user. person will be help me on the anydesk on my laptop...

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    3cx CFD & mySQL scheduled predictive autodialer with text to speech and response recognition and logging.. auto dial numbers from mysql database at scheduled time for call. (Can be many at same time), when call is answered play prompt from database (Amazon Polly TTS) may combine multiple fields or custom wave file from DB (May need to be uploaded to PBX) - Response options selected from DB ...

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    We are seeking an experienced programmer to develop and manage our Freeswitch and ASTPP billing platform currently housed on a Digital Ocean server.

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    I need a programmer to install an new instance of Freeswitch and then configure the PBX to match the PBX I have built in Freepbx.

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    i install freeswitch in windows by a public ip, but when i send calls from my softswitch to freeswitch ip that time i get a missed call, also i need to run multi freesw in same machine by different public ip and regd all extensions from 2nd ip

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    I need a 2600hz FreeSwitch instance clustered servers

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    We are looking for a developer that has used Drachtio [login to view URL] You should work to make some changes to an existing Node application that uses this and Freeswitch. This will be long term relationship. Let’s start this project by creating the milestone per task

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    We are looking for a developer that has used Drachtio [login to view URL] You should work to make some changes to an existing Node application that uses this and Freeswitch. This will be long term relationship. Let’s start this project by creating the milestone per task.

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    We are looking for a developer that has used Drachtio [login to view URL] You should work to make some changes to an existing Node application that uses this and Freeswitch. This will be long term relationship. Let’s start this project by creating the milestone per task.

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    We would like an instsane built for VoIP comany. FreeSwitch

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    In the open source code called Freeswitch, there is a code based called tone_detect and mod_avmd. I need to modify it to be able to detect a certain sound signal. This project should be easy to do. I will provide knowledge of FS and environment setup, and sample test bed.

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    I want configure a OpenSips or Kamailio SBC for load balance with mu freeswitch servers in a NAT (Amazon AWS). The users register can be use a mysql database.

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    Hi, I want create a Kamailio Server for use how SBC. The structure is: Kaimailio SBC 2 (IP 2) Kamailio SBC 1 (IP 1) Connecting on: > Freeswitch 1 > Freeswitch 2 > Freeswitch 3 > Freeswitch 4 and a lot of more freeswitchs. Need be a load balance will direct calls only for active freeswitch servers on pool and redirect anothers users to active freeswitch on a server fail. The Frees...

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    I have source code of working ios voip softphone app already. But cannot make it work with push notification (to receive calls in background), which I want to integrate with Freeswitch

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    we are getting inbound to the Local numbers, not going to the extension but doing a REDIRECTION_TO_NEW_DESTINATION to 011 numbers (we never set any numbers ) we need help on this issue

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    I am looking for FreeSwitch developer with experience dealing with mod_spandsp.

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    I have PBX with 1000 Extension. I have problem and i need change system by FreeSwitch.

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    hello i just install astpp billing system with freeswitch in my debian 9 vps. i need just bit help to understand how the rates and call plan create.. i was using a2billing before but i want to learn this billing system for some project. if you use and have proper idea how to use the astpp then please apply for the job now

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    I need to set a call limit cap on Ring Group's/ Incoming calls with the below parameters: Daily call limits Total call limit concurrent call limit People who are experienced with Freeswitch and FusionPBX can bid.

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    i need a asterisk , vicidial , freeswitch , crm expert developer , i need them for long term also to work some hours for day , please bid with you monthly sallary ! thank you !

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    Hi , My system contain: - Freeswitch server - Sip Client: Web using sipjs , mobile react-native using [login to view URL] to receive call. My problem is when call done i need to know the uuid of CDR recently add to Postgres DB of that call to attach some info to that call I try many way but can not success like , write http request to select into postgres DB, but can not find exactly which uu...

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    Hi, My project need to install kamailio to proxy for 2 node of fusionpbx using DB Postgres. I need: - install in our real system - transfer docs how to step to step install and config looking for someone who have experience about those platform

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    Hello, if you have ever setup and used FusionPBX UI to manage Freeswitch then, I need your support to solve two issues. (i) When we connect SIP server via Linphone, we are getting chat duplicates. I know its very standard on this server for which I need your help to fix it. (ii) How to enable push notifications on FusionPBX (Freeswitch server). message me if you know what i am saying. This...

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    we have implemented ASTPP with multiple freeswitch for our work, everything working fine but sofia_contact is not able to detect SIP user registered on another freeswitch and inbound calls are failing because of this. Anyone who has good knowledge of this issue contact me.

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    I need a routing configuration for the following scenario, solution could be either in asterisk or feeswitch. 1- receive sip call from calling party 2- have two sip outbound trunks, attempt first trunk, on 183 ringing (before 2nd ring) hangup called party channel (1st outbound trunk) but keep on ringing (park) calling party channel and dial second outbound trunk. 3- bridge calling party channel t...

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    We're looking for Mozilla DeepSpeech Expert to join our team. MUST have experience with DeepSpeech. If you also know FreeSwitch - that's even better,

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    Need someone who is ready right now to help with FreeSwitch configuration to user domain name in From and To value in SDP offer and SDP response using Telco SIP connectivity. Please bid fast.

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    Looking to upgrade my SIP phone setup to the latest stable versions. Using Freeswitch and FusionPBX on a Digital Ocean VPS. Currently running: CentOS 6.5 (x64) Freeswitch FusionPBX Version 3.6.3 I would also like to tune a few features that were not implemented fully during the initial rollout. I am estimating 8-16 hours, but we can talk about schedule/hrs and decide on what is appropriate. S...

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    We have several projects this being one of them. We need to modify module unimrcp for freeswitch so that it sets a channel session variable of the Unique identifier for the Channel created with the UNIMRCP server. We believe that the variable for the session with the MRCP server is in variable called `message->channel_id.session_id`.

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    I have a basic application framework that has been developed in Golang but I am missing some things that I need completed. This application framework will be used to start jumpstart application development for future applications as well as to use as a set of core libraries in existing projects. You will be working with at least one of my current developers on the project but likely more in the ...

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    I want use features of freeswitch on Astpp (for default, we cannot use the dialplan files and we don’t have the features how transfer, queue, etc on Astpp). You can make the features on dynamic dialplan or use dynamic ans static dialplan simultaneously.

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    Hi. I want the my Astpp server use dynamic and static dialplan symultaneos. I want use static dialplan for pbx functions (transfer call, IVR, etc) and dynamic for make calls.

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    Hi, I want my freeswitch with ASTPP has a transfer function when I dial a *2 and *1. Example: I receive a call. I type *2 Prompt say: Type the extension or queue to transfer I type the extension or queue number The system transfer the call (I am on line) I call to another extension or queue I drop the call The caller are transfered to extension or queue I receive a call. I type *1 Prompt say: Ty...

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    Hi, I use PBX FusionPBX and Freeswitch, everything working ok. but problem is when app in phone exit, app can not wakeup to receive call signal. I want integrate push message to ios/android device to wake up the app and receive call

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    Setup Session Border Controller Kamailio with Fusion/FreeSwitch Hosted PBX Requirements: • Kamailio SBC experience and skills when working with PBX • Understanding of Fusion/FreeSwitch PBX working with Kamailio • Understanding of SIP registrations and load balancing via Soft Clients

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    Hello there! We are currently facing a problem with our PBX system, Fusion PBX, which is a GUI for Freeswitch. It is similar to 3CX or AsteriskPBX. The problem we are facing is that users can not click on

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    hello, We are looking for freeswitch or newfies developer or expert we need AMD (Answering Machine Detection). Thanks

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    To setup and secure security and bug error on my freeswitch server

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    I'm looking for help in setting up the most basic of FusionPBX installations. I'm coming from FreePBX and would like someone to spend time giving a demo of the interface, etc... Aside from rules, here is what I have: - Want it on AWS or Google - Want OpenVPN in front of it - Integration with LDAP for Extension Information (can they auto create) - Ring Groups for Main Line - Voicemail ...

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    We are a small English company based in the North East of the UK. developing bespoke Telecommunication SoftSWITCH and Billing modules, to integrate with the third parties existing software for VoIP/SIP phone system with many rich features and is suitable for small to the midsize contact centre. SoftSWITCH - We are interested in welcoming a person with a profound interest in Telecommunication S...

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    I need a way to do fax broadcasting via fax over IP (T38). Therefore, I need someone to put together either an Asterisk server or a Freeswitch server which can broadcast faxes using whatever existing fax broadcast software that would work best with the server. I have an account with [url removed, login to view] which I'd like to use but I am open to getting an account with whatever wholesale...

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