Freeswitch voicexmlprojekty

Qualifications: - Asterisk 1.6.x PBX - FreeSwitch 1.0.x (current) Details: We need configuration and dial plans setup for Asterisk 1.6.x and FreeSwitch. For each specified dialplan, we will need the configuration for Asterisk and FreeSwitch. The configurations and dial plans are: DIALPLANS -- Round Robin ACD Call Queue (done in ael for Asterisk) -- Click to Call from a webpage -- customer types in number, it will ring the customer number and connect to ACDQUEUE -- HTML front-end needs to be done as-well (can be very basic, to test functionality) -- Click to record -- agent clicks a button and call begins to record (filename: agent-XXX-${TIMESTAMP}-${CALLERID(num)} -- HTML front-end needs to be done as...

We need an open source based hosted / multitenat pbx solution based on freeswitch or asterisk ,

Will have various projects person must have in dept knowledge of freeswitch

Doea anyone out there know Voxeo or voice xml? I have several applicationa that I inherited a project and several of the scripts work, but the attached one does not. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the natur...the platform(s) specified in this bid request. 3) All deliverables will be considered "work made for hire" under U.S. Copyright law. Buyer will receive exclusive and complete copyrights to all work purchased. (No GPL, GNU, 3rd party components, etc. unless all copyright ramifications are explained AND AGREED TO by the buyer on the site per the coder's Seller Legal Agreement). ## ...

I am looking for an experienced Voip Developer in PHP. Application should support as bellow: --Authentication methods.(user ID / IP / MAC or ANI) --Internal calling user to user + International Calling . --Alias with DID Maping --PSTN ( over SIP)calling/ billing --DID forwarding --Call forwarding --Call forwarding on busy --Call forwarding on no answer --Voicemail- Simple Text Reader +(translation). --PIN request. --LCR(Least Cost Routing) / Carrier route --Enum Support --Video calling should support. --NAT traversal support (RTPproxy, Mediaproxy or same like SBC) --Text Messaging --Webphone --Callthrough --Click to Call (Callback) --IVR (Tell the Balance every time user call). --Web Call Back 1 Customer enters source and destination numbers in Click to Call bac...

I need a software for running in a call center. All the users will connect through soft-phones, but it should be possible to support Cisco VOIP phones in future. Major requirements are 1. Full support for soft-phones. I am open to suggestions. 2. It should also support VOIP gateways for outbound calls. 3...for soft-phones. I am open to suggestions. 2. It should also support VOIP gateways for outbound calls. 3. A basic CRM is also needed, where a log of calls can be maintained. 4. The system should provide a way to schedule calls. The scheduled calls have to run against a IVR. I will provide the scripts. The data from the automated calls has to be saved in a MySQL database. I would prefer using voicexml for this, but I am open to any new suggestions. Please PM me for any mor...

We're planning to replace our current analog phone system with an ip based system. I'm not terribly knowledgeable when it comes to this area so I'm looking for a knowledgeable person that will work...in the future. 2.) The ability to to make calls on a windows machine using a softphone that can be dialed using our proprietary vb software application. 3.) Must be able to record all calls, preferably in mp3 format. The recorded calls will be ftp'd to another server for storage. We need someone who is experienced in setting up an ip telephony system. It isn't necessary that it be a FreeSWITCH system, if anyone has a better suggestion to meet our needs please feel free to respond. For the right person this would be a long term project as there are plent...

I am looking for someone to build a Flash-based VoIP widget (SIP User Agent) for me. It should be able to make a connection to my FreeSWITCH PBX (e.g. with SIP, but other protocols are with okay) and place outbound SIP calls as well as terminate inbound calls. In otherwords, a very simple Flash SIP phone - the configuration (server, username, password) will be provided programatically. Bonus points (extra $50) for TLS support to encrypt the voice stream. ## Deliverables The widget should be fast-to-download (ideally under 100KB). It is acceptable if Flash 10 is required. The UI should be very small/minimalist. Indeed, the non-debug interface should be as simple as a single button that says "join call" that turns on the mic and adds a user to an existing conference (in-...

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I'm looking for someone experienced in Sipx(sipxecs) and/or freeSWITCH. Our aim is to use Sipx as front end application to use as call center type interface by using freeswitch as media if possible. The reason to use freeSWITCH is to have a dialer type functionality working with it to make outbound calls. Currently we already have hosted based Dialer solution and looking to migrate. We already have Telephony Database & Web infrastructure where we use MySQL DB and Apache servers for configuration interface and using a proprietary telephony dialer application. Please respond if you have worked on these platforms. If you have experiance in this area and have an opinion on a better solution also feel free to bid your solution. If we find rght candidate it c...

...provider of our choice. The phone is implemented as a java applet/application and it is completely independent platform running on webpages, windows desktop, MAC, Linux, Solaris and mobile devices. It can be used as a normal softphone running on our website or as Skype-like buttons (Click to Call). Features: SIP and RTP stack (compatibile with standard VOIP servers like Cisco or Asterix, FreeSWITCH, etc) Standard java applet (no installation required. runs directly from browsers) Standard G711 codec’s (PCMU and PCMA) and speex narrowband DTMF (INFO method in signaling) Basic call features IM (chat) capability based on SIP SIMPLE protocol Other VOIP related features will be coming soon (call forward/transfer/mute and more codec’s) Encryption capabili...

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I am looking for someone with FreeSwitch For WINDOWS experience. We just want to get a feel for FreeSwitch and setup some basic functionatlity initially. Several things will need to be provided. 1.) Setup and Compile FreeSwitch on a computer at our location (can be done through radmin or pcanywhere). We have VS 2008. 2.) Setup compiled version of FreeSwitch to utilize 1 4-T1 board and 1 8port analog board, 1 IP Phone (Sound Station IP 6000), and some type of SIP connection to the outside world. 3.) Have a basic call flow i.e. call comes in from any port have it? run a IVR type script with option of ringing IP 6000 phone. 4.) Setup needs to be completed with how it is done and why (my education is critical with this project) Person will need t...

...Recipient is asked to insert Access Password. Authorizing Recipient can download the file. Configure a DID with concurent 20 incoming calls. Once the recordings have been downloaded,it should auto delte the it in 7 days. Please Note : Bids without past experience or reference will be rejected. Cheers..!! Mark Please Send us link or summry of your past expereince on Asterisk or Freeswitch....

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I need FreeSWITCH installed with ASTPP and configured. I am running Ubuntu 8.10. Installation will be done remotley so you never have to leave your house. If installation and configuration is successful there will follow up additional work.

Systems Administrator for VoIP platform including Asterisk, OpenSer, Freeswitch. Test with other carriers working as a team verifying call routing. Develop and clean up dial plans, test existing setup etc. Test existing Asterisk/OpenSer configuration, Create or Modify Asterisk dialplan and OpenSer configuration.

...assist in the customization of Freeswitch on a Linux system, with a few softphone endpoints, a Linksys ATA, setup simple IVR menus, 3 mailboxes, and dialplans. Currently freeswitch (trunk version) is installed along with WikiPBX on a CentOS 5.2 box, with a couple internal endpoints registered. There are intermittent problems with audio, and inbound/outbound dialing ( trunking) needs proper configuration. Preference will be given to those with Python skills, and experience configuring various freeswitch modules (eg: dingaling, pocketsphinx, cepstral, etc.) for follow-on projects. Ideal person would be available after project completion for consultation or further modifications, at hourly rate. Please bid only if you have direct Freeswitch experienc...

...offer our clients a hosted ITSP solution. We would like to implement the following VPSs on OpenVZ containers with High Availability and solid performance -Elastix for Hosted PBX and Call Centers -A2billing for Clients looking to start Calling Card Companies, all features with Custom GUI (look and colours and Logos) Sign Up module and Customer Interface that intergrates with Payment Processor API -FreeSwitch and a billing software for us to manage minutes charges across our client base. Maybe for simple IVR and Virtual PBX Customers We are open to other suggestions as well. For more info please contact us. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-...

We currently use elastix and a2billing on VPSs to offer our clients a hosted ITSP solution. We would like to implement the following VPSs on OpenVZ containers with High Availabi...clients a hosted ITSP solution. We would like to implement the following VPSs on OpenVZ containers with High Availability and solid performance -Elastix for Hosted PBX and Call Centers -A2billing for Clients looking to start Calling Card Companies, all features with Custom GUI (look and colours and Logos) Sign Up module and Customer Interface that intergrates with Payment Processor API -FreeSwitch and a billing software for us to manage minutes charges across our client base. Maybe for simple IVR and Virtual PBX Customers We are open to other suggestions as well. For more info please c...

I need an answering machine detection for FreeSwitch to integrate with the IVR

We are looking to create a portal for conference services. This bid is for the backend development with a placeholder UI. We will then create a separate bid item to create a nice looking UI. The person bidding on this needs to have experience with FreeSWITCH or SipX (sipxecs) as it will be the back end we will utilize for the conference services. If you have experiance in this area and have an opinion on a better solution also feel free to bid your solution. We will model the functionality after www.freeconference.com. We already have the infrastructure in place to complete the calls inbound/outbound that we can connect to. ## Deliverables Free free to ask any questions as needed. We have several projects along these lines we are looking for people to assist with.

I have one Voip Phone APP on...pjsip () I guess the app are fine , i want to work with SRTP , TLS/SIPS and the key excange are the SDES By the way, i think the APP are fine cause when i connect the app on "' or '' i have good conversation and check in logfiles that´s everything goes fine . But when i want to use in my Asterisk server i have an 'one way audio' problem , in freeswitch some kinda of 'unknown media type' and stuff. So, i can shoot that are problem with my server. By the Way i need specifcly that´s my server works correct with SRTP-SDES and TLS-SIPS . I have in my server Asterisk and freesitch already Instaled. In first moment as related above is just an server correct ajust. Will Be Waiting The Bids Mario-B...

Using either Asterisk or FreeSwitch and libraries such as Skypiax, I need a server that upon creation of a new Skype (i.e. skype username and password) user in MySQL DB, signs this user into the server and enables me to send SIP traffic with URI something like "sip:fromskypeuser-toskypeuser at sipskypeserver dot com (fromskypeuser is the user retrieved from the DB and toskypeuser is the one I am trying to call) to this server for egress to skype. if 'toskypeuser' is not on the 'fromskypeuser' buddy list than an 'addbuddy' must first be sent to 'toskypeuser' followed by the call. Need minimum ability for 100 concurrent registered skype users on this server. Calls in opposite direction (i.e. from Skype to this server) are always rou...

...Oracle, MySQL, or Microsoft SQL Server (Express or 2005) - ALL THREE. I have limited funds, but will pay for a fixed number of hours per month until the project is completed. The outbound dialing module will place calls via SIP directly, or utilize FreeSwitch, SIPXECS, Asterisk, Dialogic-based cards, or other telecom hardware platforms tbd. The inbound module will support skills-based routing and mutliple media channels such as calls, e-mail, sms, fax, and chat. Incoming call events would be received from the PBX or soft-switch (Freeswitch, Asterisk, or any other platform), and should control queuing and routing to agents based on skills or intelligent call routing rules (ANI, DNIS, Database lookup based routing). All management and agent interfaces will be web...

Hi - We need someone to make a couple of tweaks to FreeSWITCH's mod_openmrcp - we're getting a segfault on repeated accesses to a TTS server, and there's an incompatibility between it and an ASR one. We'll provide a Windows box which is already set up as a test/dev environment with Visual Studio 2008; you'll have access via RDP. --Dave

I am looking for a developer that can configure Freeswitch for me on a CentOS 5.1 machine. I will require a GUI to control the machine along with billing. The server is a Quad Core and should be able to support up to 1500 or so simultaneous calls or more. This is an urgent project as my systems are currently down and I would like the work done ASAP. Thanks in advance for being respectful of the time crunch.

We have just begun to use the openVXML Studio Eclipse plugin to develop a voiceXML application on Nuance NVP.? We have built a simple prototype using the tool and have run into a number of issues when deploying on Tomcat using Nuance NVP as the backend. We need someone familiar with openVXML? to look at the prototype and determine the source of the issues.? There is one vxml property that Nuance does not accept and audio files are not being served up correctly. We'd like a developer to help us solve these issues as well as help us for a maximum of 8 additional hours with ongoing support and debugging to help us get up to speed with the product.

A VoiceXML or Asterix application with a LAMP web site front end for user interaction. Allows a user to add customer's contact info (name, address, phone etc.) and an appointment time when to call the customer contact. When such time arrives the system will dial the user to remind him of the appointment, playing back a pre-recorded memo, and allow him to transfer the call and record a memo.

Need to develop a VoiceXML application in which: 1. user calls and request information type (already implemented) 2. for the specific type user needs to answer 1-2 questions 3. a request is sent to server 4. Parse the XML reply from the server and voice the items listed in the reply. Another similar application can be used as a reference and some components may be reused. Previous experience with VXML implementation is required. Detailed spec will be provided following NDA.

This project is the build a Freeswitch GUI. Freeswitch is the next generation of asterisk. If you have Freeswitch experience and can code in languages such as XML, PHP and Mysql this maybe the project for you. ## Deliverables The system would need the following features multitenant call routing sip endpoints gateways setup (sip, iax, zaptel) SLA Ring Groups Pickup Groups Conferencing Parking Queuing Paging and Intercom IVR Day/Night Switch Time Conditions

This project is the build a Freeswitch GUI. Freeswitch is the next generation of asterisk. If you have Freeswitch experience and can code in languages such as XML, PHP and Mysql this maybe the project for you. The system would need the following features multitenant call routing sip endpoints gateways setup (sip, iax, zaptel) SLA Ring Groups Pickup Groups Conferencing Parking Queuing Paging and Intercom IVR Day/Night Switch Time Conditions

Need a Voice Portal (voice browser) developed with Microsoft Speech Server 2007. Please provide more information regarding you skills with .Net, Visual Studio, C#, SQL, Microsoft Speech Server 2007 and VoiceXML. Once your talents and skills have been reviewed, we will gladly send Flow Chart and Specificaitons to obtain fixed bid.

Dear, we need a fully Funtionable Freeswitch/WIKIPBX Installation. For more informations please contact. Best Regards

I am looking for an experienced openSER/Mysql/php/freeradius/mediaproxy/SEM/C/perl/XML programmer to undertake development in my VoIP-portal/ Switch. This allows users to sign-up, register their phone, make calls with other registered users and to PSTN. I have SER/MySQL/PHP components but doesn't work well and I need a new openSER installation and configuration for services listed below. This bidding is for the installation/configuration from all softwares you are needed incl. additional scripts. You should make the adaptation in my existing website and if you won't finish the whole project, you should provide my PHP programmer with specifications and information. Only coder with previous experience on openSER/asterisk are accepted. 50% of the payment will be g...

FreeSWITCH's mod_event_socket claims to be able to pass bi-directional audio over a TCP connection ( - search for "bi-directional"). I am looking for example code, preferrably in Java, that can demonstrate this capability.

Assist in the development and proofreading/editing of a software architecture and development plan that includes internet, virtual reality, telephony (open source), convergence and high levels of diverse network integration. Use the vocabulary of SCA (Service Component Architecture) at the high level. Class definition and method/functional prototypes not required, but Composite a...appropriate language for the integration of the particular pre-made app. Development of a custom ESB (Enterprise Service Bus)(at a high level) for integration. Familiarity with SOA, XML, OSGI, JBI design concepts is a plus. Familiarity with Solaris and Sun Microsystems products also a plus. Familiarity with open source initiatives, especially in telephony e.g. , Asterisk, OpenSER, FreeSwitch etc.

making a application base on freeswitch. element inculde, inbound, dialplan , outbound function. This project is for voip expert with programmer skill.

I am looking for either a Java application (or Perl script but my preference is Java) to parse an XML files that will be <a href="">SRGS</a> format and render a simple text list of all possible combinations of phrases and words specif...useful feature. With regards to the SRGS standard, I will only be using XML form grammars. ABNF format can be completely ignored. The output should be regular text suitable for processing with *nix utilities or importing into Microsoft Office applications. The use of 3rd party components (under GPL, ASF, GNU etc.) is perfectly acceptable if they are appropriate. Knowledge of VoiceXML may be useful background. The intention of this utility is to test grammar coverage. Thanks for looking - Robert.

WikiPBX is an Open Source GUI front-end for running a PBX based on FreeSWITCH. It is written in Python and is based on Django. WikiPBX currently ships with a very simple voicemail system, however it needs to be upgraded to instead use the voicemail system that ships with FreeSWITCH (mod_voicemail). The two phases of the project: * Enhance WikiPBX to return the correct configuration so that mod_voicemail will work * Add GUI support so that existing voicemails can be viewed and deleted over the web interface. You will need access to your own Linux box (preferably Debian/Ubuntu, but other distros should be fine) so you can install FreeSWITCH / WikiPBX. If this goes well I will be able to feed you more work.

Simple Outbound IVR application * We are seeking a "hello world" or "very very simple" telephony application on an outbound IVR platform via Freeswitch which is a newer/better variant of asterisk * The basic "call" application is to call a given US phone number and extension at a given time, and play an audio file (mp3 preferred, or wav, or ?), based on a db entry. The call could be of up to two hours in length. (We are calling people for an audio course ... long story but there's a reason that simply web/inbound playback doesn't work for this.) * The basic "ivr" application is also very simple, and should be on the same platorm. We have a sequence of audio files , , , and again the system looks at a db

...mirror the main server twice a day and should automatically trip over if a fault should occur , and warnings sent to the engineers. Once the fault has been sorted the ability to switch back to main server should be a simple process. This is essential. The option to add video calling at a later stage with billing in place for this. All based on open source technologies either asterisk , or freeswitch e.t.c. Hosted servers will be used , and as such full control should be remote accsessable. Full Training to be provided on the system to engineer. A full list of system requirements to be sent before software is written as well as heavy load testing. System must have webpage intergration of every aspect of the system , to enable easy installation of additional operators o...

...aspect of the project. Overall Configuration: Receives and makes calls via SIP protocol. Calls actually originate on PSTN/TDM but they are transparently converted via a Media Gateway to SIP (inbound and outbound). System will use only standards-based, non-proprietary technology including: VoiceXML 2.0 speech/IVR media VoiceXML 2.1 extensions CCXML 1.0 call control SRGS 1.0 speech grammars SSML 1.0 speech markup SISR speech semantic interpretation SPEECHSC MRCP Draft ## Deliverables We will be using a VoiceXML/CCXML platform and we're currently leaning toward Voxeo but Gensys and some more obscure companies are possibilities for actual deployment. To simplify matters, all development can be done using the Voxeo Prophecy platform which is free for a 2-port dev...

I need an IVR system that will take phone orders and also process credit card payments using Internet Secure Gateway. I have the call flow already designed. The bidder should be able to implement this fully (hardware and software). Please see the attached call flow excel file.

Hi we require someone who is very experienced with VOIP. we are a new start up firm and require you to setup a system similar to (voip which works on mobile/cellphones), we need you setup everything . You can use many of the opensource systems out there: these are example Major Components Required: OpenSER:Open Source SIP server Asterisk: Open Source SIP PBX FreeSWITCH: Open source software telephony application Other Components which may be required: Apach Tomcat Servlet Container MySQL Databases Debian Linux Build Spring, Hibernate, JSF, Shale - Clay, Acegi for the web site build Roller for blogging Nagios and Monit for status monitoring OTRS for trouble-ticketing Jira for bug tracking remember you need to be really experienced in this field in order

I am working with a project to test a simple Voice app for my business. I am using a proven set of scripts that can be found at <http://www.voip-info.org/wiki/view/Add+Voice+Recognition+to+Asterisk> These two scripts, one is VoiceXML amd the other is PERL. The VXML script works like a charm, but the PERL script bombs with the infamous PERL 500 error. I need a PERL hotshot to look at my version of this PERL script and tell me what I am doing wrong. I do not think my code is bad because I have only changed the password info as indicated by the article above. I think the problem is in the way that I am setting up the PERL script. I know nothing of PERL so I can't trace the error. I need somebody who can. I will need someone with access to their own server...

We are seeking a raw Asterisk, SER, FreeSwitch, or Yate Coder, to build the internet user interface, and all the sofware and hardware operating functions to begin operations as an Internet Telephony Service Provider (ITSP) using either SER, FreeSwitch, Yate or raw Asterisk. In the first instance, the ITSP model will manage a minimum 100 simultaneous calls with the ability to grow incrementally. We are looking to create the Business model based around the Skype out model, and need someone with the ability to build the ITSP software from Asterisk, or any other underlying technology. Remuneration negotiable, and may be either directly for work done, or with possible shareholding in the entity. Disclaimer: Nothing in this bid request on [][1] constitutues a formal offer

We want to implement a very basic voice application based on voicexml (vxml). The application would be hosted by a provider (). The flow is very basic: - User calls in - Voice App collects phone number and recorded message - Submits request to backend servers - Server process the request and sends a response to the Voice App - Voice App plays a message to the user based on the content of the server response. For this project we want a working demo using the voxeo hosted solution (or if you have a different provider please let us know and we'll investigate) and a very basic server side implementation. The server side can be super trivial (and hardcoded). What we really need is the Voice App part and the handshake between the VoiceApp and the server side. After we release the fir...

Overview This is a voice autoresponder system. Instead of sending out email messages, the system calls subscriber phones and either plays a message or leaves a messages. Like an autoresponder, the system can be set up to call and leave messages "n" days after someone signs up for a campaign or it can send voice messages on explicit dates and times (adjusted for time zones). Requirements The product must interface with Joomla and user a selected vendor's API (). The Joomla interface will use custom fields that are defined in Community Builder Extensions or additional user fields (phone, time zone). Access to campaigns shall be done by reading JBAM groups (this is a system for managing user groups in Joomla) Goal Our goal is to write a custom Joomla component ...

Decently simple VoiceXML phone interface needed. We already have: - vXML Hosting () - Database in place - Exact spec for flow of the call / menus We need your Voice XML expertise in programming the interface and making suggestions for caching & performance enhancements.