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    1,741 freeswitch voicexml proiecte găsite, la prețul de USD

    We are looking for someone who is experienced with installing freeswitch, GUI interface Lua, Lua-mysql and doing all needed sip/freeswitch configuration. Install and configure: 1) latest version of freeswitch, make all maximum performance config, config sip and all other needed configuration. 2) Lua 5.2, set all directories and paths to work with freeswitch to design and compile Lua IVR applications. 3) latest version Lua-mysql 4) latest version of Blue Box GUI interface Do recommended freeswitch performance test Install and compile simple IVR application as a test.

    $285 (Avg Bid)
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    VOIP Project S-a încheiat left

    Hello, and thanks for looking. I am looking for a developer who has experience with proven results using the following to build telephony platforms: OpenSips, Kamailio, Freeswitch, Asterisk, SEMS, VoiceXML, A2Billing, GNU Telephony, etc.. Or, If you have skills using 2600hz - Whistle, Plivo, Sipwise, etc.. Thank you.

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    I need a ruby appl...application that utilizes EventMachine to handle multiple outbound client connections, and divides up the workload to the available client connections. A good candidate will have worked with EventMachine before handling multiple outbound sockets. The clients will connect to multiple FreeSWITCH machines using its Event Socket interface and initiate commands to the FreeSWITCH machines in a round robin fashion of available FreeSWITCH boxes. There should be an option to reload the list of FreeSWITCH boxes using either a HUP or database read every 5 minutes. This is a gateway project, to see other Ruby and RoR projects I also have available related to my Predictive Dialing application. I need a 100% money back guarantee from a...

    $30 - $50
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    Thank you for looking at this project. I have an existing voicemail telephone service (legacy equipment) that needs a new platform for the future. I think Opensips and Freeswitch looks the most promising. However, I know that there are many other worthy alternatives. The service must reside on a cloud provider. Web integration is part of the project and I have attached a schematic drawing that lists my needs and would be the start of a good customer user web interface. Don't forget SEO. Integrating my current SIP DID provider plus others, as well as Paypal/Google Checkout is also needed. I currently have quite a bit of PHP code on my existing site for selecting numbers from my providers database and payment/service notification. Automated provisioning for new ...

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    Thank you for looking at this project. I have an existing voicemail telephone service (legacy equipment) that needs a new platform for the future. I think Opensips and Freeswitch looks the most promising. However, I know that there are many other worthy alternatives. The service must reside on a cloud provider. Web integration is part of the project and I have attached a schematic drawing that lists my needs and would be the start of a good customer user web interface. Don't forget SEO. Integrating my current SIP DID provider plus others, as well as Paypal/Google Checkout is also needed. I currently have quite a bit of PHP code on my existing site for selecting numbers from my providers database and payment/service notification. Automated provisioning for new ...

    $1516 (Avg Bid)
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    Project – OpenSIPS, FreeSWITCH Deployment for VoIP Service Provider We are looking for an experienced programmer(s) to develop and deploy a redundant, scalable OpenSIPS/ FreeSWITCH system to be used to provide SIP trunking and multi-tennant Hosted VOIP service to small businesses. The system must support real-time communications between OpenSIPS and FreeSWITCH servers in multiple locations so that a failure of one server, or one data center has the least possible impact on active calls. Our goal is to have servers running in three separate physical data centers on Day 1 and grow that number over time. The system must support the easy addition of additional servers to support growth and improve call quality. We have looked at the Whistle app from www.2600hz.or...

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    SIP Phone iPad App S-a încheiat left

    We are looking for an experienced iPad app developer; Short list of main features: SIP VoIP calls using data and not just WiFi Offline / Online messaging (Other VoIP clients do not support offline messaging) Messages of 3 types: text messages, voice messages, photos (from camera and gallery) Contacts synchronization Inv...of main features: SIP VoIP calls using data and not just WiFi Offline / Online messaging (Other VoIP clients do not support offline messaging) Messages of 3 types: text messages, voice messages, photos (from camera and gallery) Contacts synchronization Invitations to other users Find friend by DTN (internal PCN account number) Click to call from Outlook Server-side installation (FreeSWITCH as VoIP server + Mail server application for offline...

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    iPad SIP Phone App S-a încheiat left

    We are looking for an experienced iPad app developer; Short list of main features: SIP VoIP calls using data and not just WiFi Offline / Online messaging (Other VoIP clients do not support offline messaging) Messages of 3 types: text messages, voice messages, photos (from camera and gallery) Contacts synchronization Inv...of main features: SIP VoIP calls using data and not just WiFi Offline / Online messaging (Other VoIP clients do not support offline messaging) Messages of 3 types: text messages, voice messages, photos (from camera and gallery) Contacts synchronization Invitations to other users Find friend by DTN (internal PCN account number) Click to call from Outlook Server-side installation (FreeSWITCH as VoIP server + Mail server application for offline...

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    This is a multi threaded application coded in Lua, also using FreeSwitch as an advanced IVR for telecommunications. The basic code is complete, we need someone who is really fluent with arrays, dictionary, global variables etc. We have an old application that worked well, that was done in visual voice, we can use this as a template. We have looked at the project and think there is approximately 40 hours of work to be done.

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    Hi, I am looking for someone with experience in freeswitch and especially using the mod_event_socket in outbound mode. I need someone to build me a skeleton library wich handles my events (thread per call). I will specify what functions I will need in this library, you only have to build them. The main application (server) needs to call some stored procedures or call a webservice to get some data and based on response a IVR script needs to be run. (prefferably in a separate dll, so I will have one dll per IVR script, but they will share the function library) Example functions I need: void PlayFile(string uuid, string fileName), to play a voiceFile void PlayWait(string uuid) wait for all playing media to be finished before continueing (dtmf events still need to be catc...

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    Hi, we are small internet provider, and we are looking for people to help us install a voip server using freeswitch and fusionpbx as a webGUI, but we don't want to use asterisk. You'll get a machine with public ip and one of the distros (ubuntu server 10.10, debian 6, centOS 5.5 - choose one) remote access via SSH You have to 1) Install freeswitch and source using GIT. 2) Install anything else to make the freeswitch app run. 3) Install GUI interface. 4) After successfully installed and configured freeswitch app, you'll get test login and password so we can test that everything is working correctly Are You interested ? Greetings acesabre

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    experienced freeswitch developer needed for 45 hours of dev. Requirements 5 years freeswitch experiences linux centos mysql php python voip We can discuss details more in PM. 45 hours of work

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    Goal of project is to redevelop ASTPP Voip Billing GUI...ASTPP existing functions must be 100% ported to PHP 3)ASTPP supports Asterisk, Freeswitch and openser, please keep this in mind when developing the GUI.(As new GUI should support all functional softswitches) 4)ASTPP supports third party applications example: FreePBX, Thirdlane, Fusionpbx and osCommerce 5)We would like to develop support for: Bluebox and Whistle from Please note GUI should be develop in a modular design to support third-party plugins. It must use some sort of PHP framework to develop GUI. We will provide Administrator template which will need to be customised using mvc framework, jquery and ajax. Template that we provide will have all required widgets. Please note Freeswitch integration is priori...

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    Replace existing Java client with simple ESL client to fix memory leeks on freeswitch system.

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    Seeking a freeswitch developer for 100 hours of work for bug fixing and trouble shooting a existing FS installation: 5 years required Freeswitch Linux CentOS mysql We can get into more details in the PM

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    Przedmiotem zlecenia jest przygotowanie działającej konfiguracji routera OpenSIPS w celu zapewnienia konfigurowalnego load-balancingu połączeń kierowanych na serwery aplikacji/mediów oparte o FreeSWITCH. Dokładnie chodzi o poprawne zaimplementowanie konfiguracji OpenSIPS opisanej na stronie: Niestety konfiguracja tam przedstawiona nie działa w trybie "kopiuj-wklej", trzeba poprawić co nie działa i dodać obsługę NAT. Dla speca znającego OpenSIPS poprawienie tej konfiguracji i dodanie obsługi NAT to pewnie chwila roboty. Założenia: - Na adres OpenSIPS'a kierowane będą wszystkie połączenia (wychodzące, przychodzące) - LB kieruje ruch na jeden ze zdefiniowanych serwerów (wg wagi - z możliwością konfiguracji)

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    ...in front of my FreeSWITCH application/media servers. What I'm trying to do is described on FreeSWITCH website: Sample configuration from FS website needs to be adjusted to my needs (copy-paste does not work). The configuration is really simple, but I completely don’t know OpenSIPS and how to fix it. Detailed scenario: 1. All incoming traffic goes via Load Balancer, it has public IP, no NAT 2. LB selects destination server from one of available (online) FreeSWITCH nodes (also public IP, no NAT). I must be able to manually adjust list of those servers via database 3. LB forwards SIP request to chosen server 4. LB does not handle RTP media, media sessions are made directly between users and FreeSWITCH nodes. 5. Clients

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    Seeking a freeswitch developer with a minimum of 4 years experience to add a few features to our freeswitch based telemarketing app. Job will be 40 hours of dev requirements: freeswitch linux centOS php mysql sip we will discuss more details in pm. thanks for biding

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    We need someone with experience with the CMU Sphinx 4 toolkit, and Asterisk or Freeswitch. We seek professional real-time dictation of SIP channels to text, with the ability to continue to tune or improve quality of transcription. Requirements: 1) The system shall be hosted on Amazon EC2 2) The system language shall be English 3) The system shall allow the users to improve the basic accuracy in some way, such as asking the user to read from a known script of training words. 4) The system shall translate multiple simultaneous SIP channels. Please do not submit quotes if you do not have the prerequisite experience with CMU Sphinx, or have not done similar work in acoustic recognition.

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    I need someone to do the following: 1. Install FreeSwitch on UNIX Server 2. Program custom call flow that MUST communicate with SQL via lookups 3. Must program the call flow to UPDATE or INSERT input values from call flow into SQL associated with user/caller authenticated ID 4. Program the call flow to allow a user to record a message and update the SQL data field with unique file name of recording that is associated with the user/caller authenticated account ID 4. Assist in programming a SQL stored procedure to create a multi user call list of phone numbers that gets pushed to 3rd party VOIP dialer network. The file will include the numbers to call and any recorded files that need to be played. I can explain and answer any questions that you may have.

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    We are a mid/large company that is looking for someone to write code for a CDR call log/call recording interface that would use perl, HTML4/5, mysql and erlang to connect to mysql database and freeswitch boxes to extract and display info. We already have a basic setup but need to expand and develop furthur on it. Need perl script writiten for a web gui front end including authentication and authorization. This will be a ongoing project as items will need to be added and modified as time goes on. We are looking for someone that is familar with the voip field and has all skills listed. We will agree upon a hour range for the items that need to be accompished. As more items need to be accomplished we will agree upon the hour for those items. Just needing the code to be wrtting and docc...

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    Phase 4 of Freeswitch Telephony Dev project please bid on 60 hours of freeswitch on linux (centos) development. This is phase 4 of a ongoing project.. Skills needed: CentOS Linux Freeswitch mysql php Minimum 3 years experience. details will be provided in pm

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    Class 4/5 - SBC / SoftSwitch 1. Routing Engine 2. Billing Platform 3. Rate Engine 4. Load Balancer 5. Origination and Termination Point configuration 6. Report query 7. Automatic Invoicing System. 8. Trouble Ticket management 9. Email Server for Reporting Preferred SIP Server: FreeSwitch, OpenSIPS or Proprietary.

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    Please quote 10 days of work for Phase 3 of a freeswitch development project. We will discuss details in pm.

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    We have an IVR running on freeswitch and we can't receive calls, or forward, there is no audio but the switch claims that the call is answered.

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    We would like a Freeswitch server Stress Test ( Load Test) . We are curious how many simultaneous calls it can handle and other information. The tool used for the stress test should be SIPp. "I swear, under penalty of perjury, that I am the owner of the server in question. In no way am I attempting to exploit the security weaknesses of a third-party server for any malicious purpose. This is simply a stress test of our own personal server and is in no way a potential threat to any third party."

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    Purpose: Make a simple multi-user PBX with live call operator or voicemail answering calls. Overview: We need a FreeSWITCH installation with certain features, which is controlled through an ASP.NET interface. The job includes installing an configuring FreeSWITCH on a physical machine, and also creating a web page in ASP.NET (hosted on Azure), which controls the live calls on FreeSWITCH. Looking forward to your quote, selection and quick execution! ## Deliverables Details: 1. Make FreeSWITCH installation with the following features: a. Forward incoming call to operator's softphone, where call queue is possible. b. Receive incoming call via DID c. Forward call to one or more outgoing numbers via SIP gateway d. Forward call to voice mail e. Customize voice ...

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    Hi, we are looking for someone who know voiceXML and CCXML to join a speech app project. Please bid $30 and we will need to discuss your qualification on PM and set up the fee based on the discussion Thanks

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    Hi, we are looking for someone who know voiceXML and CCXML to join a speech app project. Please bid $30 and we will need to discuss your qualification on PM and set up the fee based on the discussion Thanks

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    Freeswitch VoIP Phase 2 of a fs platform - Linux, MySQL 80 hours of work in freeswitch, linux, mysql, and php.

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    We currently have a Switchvox AA65 in our office. We want to replace the switchvox system entirely with an open source VoIP system. We expect all parts of the VoIP system to be actively maintained by the developers (e.g. Asterisk or FreeSWITCH, /not/ trixbox). We have a dedicated Internet connection strictly for VoIP, no landlines, 11 Aastra phones and less than 10 DIDs. You will be provided remote access to a dedicated freshly installed Linux server. Your job is to configure the VoIP system and either provide simple instructions to configure for Aastra's (providing support if needed on configuration) or set up some way for the Aastra's to automatically configure themselves (e.g. a TFTP server) Required features: Support up to 15 Aastra 9133 SIP phones, each w...

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    Need some changes made to an existing freeswitch IVR code. I would also entertain the idea of of migrating to Asterisk.

    $30 - $5000
    $30 - $5000
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    freeswitch developer needed for week long project.. details will be provided in PM.

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    We provide SIP trunking and Hosted PBX services to small business. Currently our "edge" server is based on Asterisk/A2Billing. We are considering migrating to an OpenSIPs/FreeSwitch based solution if it will offer greater scalability, and reliability. A2Billing is currently working very well for us, and we like the fact that it is a "system" that has a good feature set and cohesive interface. We are bringing a second location online for physical redundancy and we'd be interested in moving to an OpenSIPs/FreeSwitch based platform. The system must support seamless failover between locations in the event of an outage. We already have an existing customer base, so we're not open to unproven systems. We are only interested in developers that ca...

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    hi, i need someone to with experience of GSM phone integration Asterisk or FreeSwitch with GSMopen or something similar

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    20 Blogs order in two main areas 1.... Please read below. PLEASE NOTE ... Toronto & Markham ... City's name in canada Let me know if you have any questions QTY 10 - Blog Keywords - NOT HEADING ------------------------------------ Business Telephones Business Phone Systems VoIP Telephone Systems Phone Systems Toronto telephone systems Hosted PBX Asterisk Consulting FreeSWITCH Consulting Hosted IP Voice Business Phone Service QTY 10 - Blog Keywords - NOT HEADING ------------------------------------ IT Support IT Support Toronto Toronto IT Support Markham IT Support Outsourced IT Services Toronto Small Business IT Services Small Business IT Professionals Managed IT Services IT Consulting IT Support Services

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    Freeswitch Telephony Developer Needed. Scope for IVR app.. please quote days and cost. Freeswitch & User interface -One single Did hosted with sip trunking will ring into the freeswitch app. -Mp3 (attached) will promoted user to enter their 5 digit zip code then # or press * to start over. -Route (forward) calls to a US phone # based on zip code entered by caller. We will need to upload a csv or tab delimited txt file with column1: zip code, column2: 10 digit phone # -Provide a call log either in the UI or in a csv download. ** this could evolve into a larger project but this is the phase 1 Requirements: Freeswitch Experience SQL server 2008 Ability to communicate by Skype or windows messenger for part of pacific time day hours 8am to...

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    ...when I try to review or call it up later it does not play. I think it may need some sort of linking from file to lams2...IDK that is what I need someone to help with. The thing is it records but I can not see the finished movie once finished. Secondly, OpenSim + Freeswitch both successfully working seperately on the same server. I presently use the freeswitch on our BigBlueButton install for voip video conferencing. And OpenSim also works but not with Freeswitch.I had another freelance pro install it so I believe the Freeswitch module for Opensim is installed correctly. So I need some one to troubleshoot the problem and solve so that we can use the voice in our opensim program. The server is dedicated Ubuntu 10.04 64 bit/ Linux server and is used primairly for...

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    ...are looking for a Technical creative article writer. We need 10 SEO unique articles, each having a lenght between 500-600 words !!All content must be WELL RESEARCHED, you must do a comprehansive research and be knowledgable in the topic.!! !! SEO Knowledge . Using Technical keywords. Quality Content !! Topics: 1. Hosted PBX , 2. Business VoIP 3. SIP Trunking 4. Asterisk 5. FreeSWITCH 6. Blackberry playbook 7. Android Tablets 8. Android Phones 9. Windows Server 2008 10. Windows 8 Beta All work must be 100% original and pass Copyscape and Plagiarism spelling and grammatical errors! We will own all copyrights to all work upon payment. We need a Test Blog. Before we award this project, we want to make sure we are hiring a right person. Th...

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    Freeswitch Telephony Developer Needed. We are talking to freelancers to build a freeswitch telephony app. We can provide more details in PM. Please bid on 100 hours of freelance work. Requirements: Freeswitch Experience SQL server 2008 Ability to communicate by Skype or windows messenger for part of pacific time day hours 8am to 10pm 30+ hours a week availability

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    Specific for a freelancer

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    Hi, I need to have a system configured for a VOIP provider. The system should consist of 2 server roles: 1) SIP Proxy (OpenSIPs) Functionality: • Act as a user-facing point of entry at the edge of a SIP network • Registrar Authentication • Failover/Load Balancing • Topology hiding (this can be discussed further) 2) by 2600Hz (freeswitch based) • Multi-Tenancy • All hosted VOIP calling features (IVR, Conference, SIP trunk etc..) • Central phone provisioning • SIP Trunking (some tenants may choose their own SIP trunk providers, some will user gloabal providers) • Integrated with billing DHT system Along with the setup, the detailed documentation/training for system administration will be required on topics such as:...

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    We currently have a single A2billing server with a few clients on it. We are expecting the numbers to grow soon. We are interested in knowing more about creating a multi-server cluster using A2billing, OpenSips/OpenSer and Asterisk/Freeswitch. The cluster should have a single instance of A2billing so that we can manage everything on one interface but the calls should be load balanced over multiple servers. We are currently capable of doing over 300 max simultaneous calls on a single A2billing server. Our target figures are 1000 simultaneous calls, so we would need whatever can give us that. The setup needs to be expansible, so say few months down the line if we need more capacity we should be able to add more servers to increase the overall capacity and thus number of simultan...

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    We are looking for a Technical creative article writer. 20 Articles ( 1 Article Per Day) Delivery on Mon - Fri ( 5 days) Topics: HostedPBX , VoIP , Business VoIP, SIP Trunking , iPhone Apps , Android , Cisco Phones , Open Source Telephony ( Asterisk & FreeSWITCH),Telecommunications IT Services ,Microsoft Windows Servers , Microsoft Windows 7 , Microsoft Office , Browser,Linux, Mac , Tablets etc SEO Knowledge . Using Technical keywords. Quality Content Length of Articles: 500- 600 words All work must be 100% original and pass Copyscape and Plagiarism spelling and grammatical errors! We will own all copyrights to all work upon payment. We need a Test Blog. Before we award this project, we want to make sure we are hiring a right person. Thanks

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    Turn MWI on SIP phone S-a încheiat left

    All, I have a custom voicemail server running using Dialogic HMP and am having issues sending MWI (Message Waiting Light) SIP Notify Messages to the SIP phones to alert them to waiting messages. If you can get MWI working with s...a custom voicemail server running using Dialogic HMP and am having issues sending MWI (Message Waiting Light) SIP Notify Messages to the SIP phones to alert them to waiting messages. If you can get MWI working with some Dialogic HMP code that would be best - if not I need some code working on Windows2003 (Freeswitch) that I can call with the user SIp URL (i.e. sip:1000@) that will turn his MWI light on. For the client side I'm using a EyeBeam Sip phone registered to FreeSwitch. Thanks - in responses please explain how you will design t...

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