Freeswitch jobs
We are a telco company and run our own voip platform. We need to develop a small application for the SNOM REDIRECTION SERVICES to handle all that available XMLRPC Commands off the service in combination with your data. To scan the barcode of the phones for the MAC address we need also to integrate a barc...server then to ours and then get the config file from our server and firmware. More information about that services and commands you can find directly by SNOM. :XMLRPC_Redirection The user interface shall be webbased If the work found satisfactory more projects like FreeSWITCH (SBC) or VOIP application development will be assigned to the same person or company.
Hello we are a new VOIP/SIP Provider and need to customize the Asterisk A2billing and also be able to add a few modules such as multiple DID Forwarding (Check Box next to DID to be forwarded. ) screenshots will be provided in email. Its ok to email me. We need to customize the Customer GUI and we also need customization using Kamailio, FreeSwitch and OPenSIPS.
hi, i have a small task to be completed in freeswitch. current freeswitch is running under NAT. and audio is not working. i need a solution for this so that freeswitch can be accessed from external please contact with me if you are available happy bidding.
The project consist of building a .Net based windows GUI for administration of the Freeswitch service. to build a windows .net based VOIP PBX. the freeswitch open source project can be found at: <> the GUI will be made with a very well known Suite of components, Syncfusion that can be found at: <>
I have a freeswitch server running, users are able to register, but cant dial no extension. Users can join the conference and hear each other. I need to debug this server, and make it work very well. Simple project, 30-60 minutes job. Any question please post.
Deployment of a Linux Based Open source VOIP server having all standard calling features. Feature List. 1. VOIP - VOIP Calling. 2. VOIP - Analog Calling. 3. Skype support for inbound/outbound calling. 4. Must be Voice-mail ready. 5. Multipa...support for inbound/outbound calling. 4. Must be Voice-mail ready. 5. Multiparty VOIP Conferencing. 6. Secure, reliable and technically rigid architecture. 7. Must support G711, G723, G729 and Speex voice codecs. 8. Easy to use and deploy. 9. Enhanced audio quality with no or almost no delay. 10. Integration with third party software's if needed [Customization needed]. Asterisk and Freeswitch will be preferred as an advantageous options. Serious bidders are needed, Post back if already experienced, A live demo will be a...
We need someone who can both configure FreeSwitch and also make some changes to the FreeSwitch source code. We need a redundant FreeSWITCH configuration (minimum 2 servers) running as a 'soft switch', aside the softswitch part we require 'n' servers running (complex) IVR scripts. The soft-switch part should have SIP trunks with multiple carriers on all these trunks we receive incoming traffic and allow outgoing traffic. The switch should query an MSSQL database to get routing-plan (ie, destination number+trunk). The IVR setup we would probably configure as a sip trunk as well so this can be a destination too? Al calls (in + out) should be written in a database (CDRS) and need to be 'carrier-grade' reliable. (meaning written directly...
Freeswitch is installed every thing working well except conference, voice is ok for conference announcement like "you are only person in this conference" but as soon other person connect voice quality get chunky, Very simple task if you have existing experience with this, so only bid if you already have fix that error Note: quick and simple job so Bids higher then mentioned budget will not be considered!!
HI , We are looking for a voice survey system using IVR , based on freeswitch or Asterisk . The system should handle multiple tenant IVR survey system and each survey should handle multiple question and hierarchy model questions . Text-To-Speech should be integrated to add surveys . easy to use , stable and should be handle as many as simultaneous calls . Reseller panel should be available and also a standard billing should be integrated in the system . Please bid only if you can show me any voip projects you handled. Thanks David
You will be given ftp and root access to server. Current configuration: 1. gateway <> router (NAT) with port forwarding <> FreeSwitch Server <> clients (wholesale customer/x-lite customer) ? The purpose is to route traffic from x-lite clients to gateway 2. x-lite client / gateway <> router (NAT) with port forwarding <> FreeSwitch Server <> voip A-Z ? This is another functionality implemented (mainly the same devices / trunks but routing calls in the other direction). The main objective is to fine tune the system that has been implemented. There is set of implementations that are urgent and others that can be scheduled not so tightly - please check info on project details. **1** ## Deliverables 1. **1. ** **Urgent** ? Assur...
This project is to install and configure asterisk or freeswitch on our PC, and create the following simple callback feature that should work with our designated carrier. 1) Create a web page that I can easily create new sip addresses and new call back numbers. So for example, I should be able to easily enter a new DID number and callback number, it should then automatically create a sip address, such as, xxxxxx@ipaddress and all calls that ring to that SIP address using a DID number, the system should automatically create a callback to the number designated for that sip address. Also create a call log for each sip address accordingly. 2) Once the user calls the sip address, it should play a busy signal, and then the system should call back the designated number as stated...
we need a FREESWITCH installation with freepbx, working conferences and voice mail, also need a way to block users from dialing eachother, we need to be able to decide who will call who. Also need the ZRTP protocol working with freeswitch, as MiTM and the regular way. just need 3 extensions configured, of course we will add more later, using freepbx gui.
I have a website and I require a feature to be implemented to the website where users can use some of the website features by calling into a number. The feature I require involves the user be able to communicate with the website by calling in. The user may perform duties such as log in t...with the website by calling in. The user may perform duties such as log in to their account by dial pad, check in to an event, and check out, etc. Example: System: *Welcome message* enter your account number -user dials account number System: Enter your password -user dials password System: To check in press 1, to check out press 2 -user Dials 1 to check in I need asterisk expert or freeswitch to work on system to be configured to route and handle the incoming calls based on defined p...
I need someone with good experience installing and integrating FreeSwitch. I want to create an install on Amazon cloud and connect my Voip application. Also looking to build a scalable dialer to broadcast thousands of recorded calls a day for non-profit organization. Please tell me your experience with FS.
We are an ITSP looking to provide SIP... The first phase requires building an OpenSips and Freeswitch/Asterisk cluster. Please see attached network diagram. The cluster specifications will be: - Scalable - Fully Redundant - Configurable (easily add new accounts and set maximum number of concurrent calls per account) - Burstable number of concurrent calls per account - Voicemail system - Call hunting - DIDs and LCR - Monitoring tools for checking peer status, ping time, rtp, network issues - Monitor critical cluster services (Mysql, Asterisk, freeswitch, OpenSips, etc) - Calls can be authenticated by username and password, source IP address - G.729 and T.38 Support Please bid on this project only if you have extensive knowledge working with Asterisk/Free...
The project consist of building a .Net based windows GUI for administration of the Freeswitch service. to build a windows .net based VOIP PBX. the freeswitch open source project can be found at: <> the GUI will be made with a very well known Suite of components, Syncfusion that can be found at: <>
Set Up a FreeSwitch server so that it will be able to * route SIP incoming calls (with user CLI) from other devices (these devices are located within the LAN and will SIP register to the FS Server and each device might send several different CLI numbers) to voip provider's Trunks (some 4 or 5 known voip operators), according to * dial plan (just about 10 alternatives); * user authentication (against CLI or username list - to decide then); * for calls comming from selected devices, FS will present dial tone and receive dtmf, parse it and proceed according to dial plan. * set up of nibble billing; * setup one or two accounts so that devices located outside the LAN (in the web) can register and make SIP calls * accept / reject SIP calls incom...
ITSP VoIP Developer sought to deliver turn-key service provider website and open-source server software. 1. Provide Admin interface - authentication, configuration, billing, service options, content management, reports, etc.. 2. Provide User interface - authentication, payments, customer self-service configuration, content management, reports, etc.. No need to reinvent the wheel. Asterisk, Freeswitch, Bayonne, OpenSer-Kamailio, OpenSips, SER, sipX, etc. have all been used for this type of application. Web development skills could be PHP, Ajax, MySQL, Jquery, LAMP, Zend, CRON, Javascript, HTML-CSS, Web 2.0, Cake, etc.. The application has these main components: 1. Obtain a DID number from a carrier API. 2. Forward this call if needed. 3. Simultaneous ring "find me&...
A configuration module cum dial plan maker for freeswitch IPPBX. Needed Urgent. Must have good looking interface.
I am looking for someone expert in freeswitch to install and configure freeswitch with all codecs with sip and h323. I understand h323 is not well developed in freeswitch but I would like your suggestion in selecting one. We will run it for PBX and wholesale. Your task would be to install and help us configure the switch. The OS is centos 5.x.
We need a FreeSwitch accustomed developer that would write a module/script that connects to a billing system API via HTTP. Basically, the script will need to do 2 things: - when a call is initiated on the switch, it needs to "ask" the billing if it can make the call, and also needs to route the call according to the answer from the billing - at the end of the call, it needs to send the CDR details to the billing system In the second phase, we will need realtime billing: the script should ask the billing system every minute if it should let the call continue, or if it should cut it We have already developed the API of the billing system. Installation/configuration of FreeSwitch on a Linux box (CentOS or Fedora) is also needed.
Hello, Looking for a developer who can install the latest Diva Distro of Opensim, also install Freeswitch voice chat for Diva Distro and C$ , currency for Opensim. Please send me your offers. Keep in mind that we are looking for a skilled professional. It's a job which will not and should not take more than 1-2 hours. Please bid logically.
Project Name: Telephony Server w/ Front-End Port Project Components: Web Front-End (wfe) - Connect a simple graphic interface to the FreeSWITCH will provide the UI for you to: (a) integrate with the Back-end; (b) port our mobile app business logic to the web. This port needs to be as close to line-by-line as possible. FreeSwitch Dialer Back-End (fsD) - Build a basic dialer, based on FreeSWITCH which implements the following API’s: startSession() - records a list of callers and callees for a session agentLogin() - accepts a call from a pre-registered caller getCalledId() - returns the callee id agentReady() - indicates caller is re...
I'm looking for someone with expertise who can modify existing Freeswitch endpoint modules and can create custom endpoint modules from scratch.
We need to configure FS IVR system with - Extension per member - Multiple DID management - multilingual - Configurable prompts based on extension and DID - IVR menu configurable by member and DID - TTS or uploadable files for prompts, some membe...and use appropriate menuing. Time Estimate 1 to 2 Days iii) Create Voice Mail Recording module that stores meta-data in a database and sound file on shared File System. After a voicemail is left platform triggers notifications to subscriber and/or agent based on Create call billing software for incoming calls. And Stub API for Agent Billing (Most of this will be accomplished Via Existing FreeSWITCH Software (mod_nibblebill for incoming calls that are billed per minute (i.e.: Toll Free Access) and additional services as may b...
Server set up and testing FreeSWITCH Setup (compile) Mod_nibble,fax,spidermoneky,database connectivity Skype, dingaling(gtalk), cdr_xml, xml_rpc
We need a freshwitch expert for various set up actions : Server set up and testing FreeSWITCH Setup (compile) Mod_nibble,fax,spidermoneky,database connectivity Skype, dingaling(gtalk), cdr_xml, xml_rpc Deploy Databases ERD onto DB server : Create Required Database and Table Structure to translate graphic ERD To working SQL Create required Developer Logins to DB Server Software Allow Access to DB From Required Remote Locations (IE: Unblock Firewall for developers needing remote access to DB Server for Remote DB connections) Combined with creation of access logins. While this is a separate step it allows us to restrict who can connect directly to the database and allows for firewalling and IP address restrictions) Create Inbound Call Handling Application : IVR crea...
...SEO content for Product Response web project There are 3 web sites providing VOIP maintenance services. We're looking for person who will create high-quality unique content for 3 web sites in accordance to our keywords. VOIP, Asterisk PBX knowledge and technical writing is appreciated. Basic SEO knowledge is a pro. Services description: Freeswitch services Freeswitch general troubleshooting – FreeSwitch Support Freeswitch billing with mod_lcr + mod_nibblebill Bluebox pbx (BB 128, BB 256, BB 512, BB 1024 and BB FREE) Fusion pbx sipXecs unified communications Kamailio services Kamailio general SIP:Provider CE / Pro Asterisk services Aterisk general troubleshooting - asterisk support (General...
We are looking for a VOIP Master for an advanced online platform that manages mainly incoming voice + data services. key features include : Shared DID numbers for members Configurable IVR for each number with specifc forwarding rule Multilingual features member can forward inbound voice to SIP, skype or gtalk or live operator Inbound fax with possible software code scan for automatic dispatch to member We also need a plateform to manage remote operators/agents that can perform various tasks for members (post task / execute task / charge / feedback / availability management ) Services and membership to be paid for with prepaid credit /billing system Detailed requirements in PM Work to be started immediately We can supply detailed DB mapping and team that manages front end ...
Hello, Looking for a developer who can install the latest Diva Distro of Opensim, also install Freeswitch voice chat for Diva Distro and C$ , currency for Opensim. Please send me your offers. Keep in mind that we are looking for a skilled professional. It's a job which will not and should not take more than 1-2 hours. Please bid logically.
...customers has a dedicated virtual server running asterisk 1.6. The virtual servers run in on a proxmox ve cluster with private IP addressing. The purpose of the SBC is to allow the remote IP phones (primarily Polycom and Cisco SPA's) to register with an SBC on the public internet which then passes the registration information along with the media to the appropriate asterisk server. OpenSIPS, Freeswitch and Kamailo all have fairly well documented instructions on how to configure them to act as on SBC for asterisk. We would be happy with any of those applications. Two servers will be used for this application and must be configured in an HA architecture (using heartbeat or some other application). The HA availability must be available to both the remote phones accessing th...
REQUIREMENTS: - You must be an expert in Freeswitch - We are NOT interested in an Asterisk system, or any other type of system other than Freeswitch - You must NOT be a part of a large organization (defined as having 15 or more employees/contractors) - I prefer to work with either individual consultants or smaller groups We are developing a feature rich teleconferencing solution. Video-conferencing is unnecessary for this project. We are developing a teleconferencing platform where a user can serve as a moderator. This moderator can invite any number of friends or co-workers into his/her teleconference. Our platform provides additional features above and beyond a basic teleconferencing system as described in the PDF document attached. We will need both the ba...
We are looking for a solution based on Freeswitch capable of doing the following main points 1. Wholesale Origination / Termination 2. Callshop module with three level of Reseller System 3. Web based access for the resellers and call-shops owners 3. Integrated Billing 4. Web based panel for wholesale clients 5. should be able to manage pre-pay/post-pay customer RATE MANAGEMENT (Billing increament, peak and off-peak rates, import rate sheet, rate sheet simulator) BILLING REPORT AND INVOICE MANAGEMENT (CDR (Admin,Agent,Customer), Calls ReporT, Profit & loss report(Admin,Agent),Traffic Report,Search Criteria for reporting,Export report to PDF & CSV, Invoice generation) These are the basic feature that we are looking for there might be couple of more but in...
Will map conf application written in LUA to run on a Freeswitch machine
I need the Astpp () billing software installed and running over "Freeswitch" () running on a server on Centos 5.5 I try but dont have the time to continue this project. Will be done remotely on a hosted server. If successful may work in further projects Any further inquire please pm me.
The project consist of building a .Net based windows GUI for administration of the Freeswitch service. to build a windows .net based VOIP PBX. the freeswitch open source project can be found at: <> the GUI will be made with a very well known Suite of components, Syncfusion that can be found at: <>
Dear Freelancer, We need fully functional SIP/jingle client completes with the UI for Desktop Windows XP/Vista/7. All features required: 1. Skinnable UI. 2. Interoperable with FreeSWITCH. 3. Supports login, presence and status, contacts, interaction with web service, text chat, voice chat, voice conference and video chat functionalities. 4. Supports NAT traversal such as STUN and ICE. 5. Supports Voice codec such as G711, GSM, Speex, iLBC and G729. 6. Programming languange C/C++ or .NET. The project will be evaluated based on: 1. Quality. 2. Support. 3. Price. 4. Time. The delivery of the project should include source code. The project will have support for bugs fixing. Thank you for your time.
Hello, I need a pro to give me a in depth understanding about a2billing, freeswitch, softswitch, sbc, transcoding, asterisk, voip, sansay. I have a basic understand but there are some questions I have and need to understand it better than I do right now. Example of a questions. What is the codec with the minimum bandwidth that will allow the receiving party who an ip-pbx ivr auto attendant to allow the dialing party to use dtmf to press the correct extension? I would like 1 hour of the person dedicated to me by chat or phone. Im flexible on the day and time.
We are in need for someone to help in writing telephony outbound application using freeSWITCH open source platform. ? I would ? like to use event socket library to talk to freeSIWTCH. This means we will run our application out of process. Our application will be in C/C++ so you must be proficient in C++ , object oriented programming. ? Also ope for other options like ErLang etc..? If you have worked on/using freeSWITCH then you have already smelled about the plan to use CentOS system and scale upto 400 channels with concurrent calls. We already have dynamic callflow for play and record and digit collection. The basic requirement is ? system needs to make call on channel to do activities like play/record/digit collection and transfer ? to other party. We must need to handle u...
We are looking to build a conferencing server using Freeswitch and toll-free / local DIDs. This project will not use Asterisk, or any type of Asterisk components/distros (PiAF, A@H etc...) It is a PURELY FREESWITCH INSTALLATION AND CONFIGURATION! Thanks, and let the bidding begin!
We are in need for someone to help in writing telephony outbound application using freeSWITCH open source platform. ? I would ? like to use event socket library to talk to freeSIWTCH. This means we will run our application out of process. Our application will be in C/C++ so you must be proficient in C++ , object oriented programming. ? Also ope for other options like ErLang etc..? If you have worked on/using freeSWITCH then you have already smelled about the plan to use CentOS system and scale upto 400 channels with concurrent calls. We already have dynamic callflow for play and record and digit collection. The basic requirement is ? system needs to make call on channel to do activities like play/record/digit collection and transfer ? to other party. We must need to handle u...
We are looking for a PHP senior expert to help on the design and production of a web application powered by freeswitch (voip service similar to tollfreeforwarding dot com) The PHP developer will work in team with voip Expert in order to make GUI including the users screens and interface with the freeswitch backoffice . this include multilingual website, registration and membership wizard, map powered directory and local agents registration and management. Voip Expert is a very senior dev and will provide API alnd technical detailed guidance. User GUI will include google map integration (geographical navigation) and will be ported on web and mobile and possibly desktop (flex) BID ONLY WITH URL PROVEN WORK REFERENCES IN BID We will require impeccable communication and an...
Looking for programmer with Telephony experience. You w...experience. You will be completing a conferencing application. Application is written, but the person who wrote the application is not available to finish (personal issues). Application runs on a CentOS machine using Freeswitch as the telephony engine, an Adtran 904 router with a Sangoma voice card. Code was written in LUA and uses a mysql database. You must have experience with TNCInet's VOIP service. You will configure the application to run with TNCInet, establish backup procedures, and make any final modifications. Must have prior experience with CentOS, LUA, Freeswitch, Mysql, Sangoma voice cards and Adtran 904 router. This application is in the Adult Services field. Looking to hire someone immediate...
Looking for programmer with Telephony experience. You w...experience. You will be completing a conferencing application. Application is written, but the person who wrote the application is not available to finish (personal issues). Application runs on a CentOS machine using Freeswitch as the telephony engine, an Adtran 904 router with a Sangoma voice card. Code was written in LUA and uses a mysql database. You must have experience with TNCInet's VOIP service. You will configure the application to run with TNCInet, establish backup procedures, and make any final modifications. Must have prior experience with CentOS, LUA, Freeswitch, Mysql, Sangoma voice cards and Adtran 904 router. This application is in the Adult Services field. Looking to hire someone immediate...
