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.
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
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.
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...
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
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 build a SIP softswitch using open source SIP stack such as freeswitch or IPTEL, and also build a SIP/Q.931 gateway, running on linux. ## 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 nature of the deliverables): a) For web sites or other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer's environment. b) For all others including desktop software or software the buyer intends to distribute: A software installation package that will install the software in ready-...
