Freeswitch sbctrabajos
Install Big Blue Button Video conferencing software on a server. Suggested host for the project. Use API to set up a conference scheduling system. Align Conference call in # via FreeSwitch Auto email out calendar invite to members of conference Then when Big Blue Button comes out with meeting record this fall, I will need to install that and make functional.
add DID's to freeswitch bluebox and set up auto attendant to answer these calls and allow to transfer to the extension of the person they wish to talk to.
This project is about getting backlinks for my website. Requirements are as follows: * 500+ PR4+ backlinks. Some websites must be PR7 and PR8 * All links must be minimum PR4 or above. * All links must be appropriate for our website (business telephone systems sales, install...address as well as the date placed. Read the above terms carefully. If you agree to terms, you're welcome to bid. Do not bid if you don’t fully agree with above terms. Keywords to use are: Business Telephones Business Phone Systems VoIP Telephone Systems Phone Systems Toronto telephone systems IT Support IT Consulting Small Business IT Services Hosted PBX Asterisk Consulting FreeSWITCH Consulting Cities: Toronto, GTA, Markham, New Market, Vaughan, Whitby, Pickering, M...
need someone to install and show me how to install 2600hz project called "Whistle"
In relation with PHP developper and shared specs, work should lead to following demo deliverables and features (member = subscriber to service) : - Member Voice mail configuration (see member panel) - Forward call for each IVR selection for each DID to voice mail or SIP or Gatlk based on member settings - Fax Inbound (stored as tiff file format and db table will be populated) - Call prompts are based on each DID default setting or member selected IVR (example 1 sales etc.) custom prompts (TTS, wav loaded by member) for each DID - Caller ID added to member directory (or caller promoted to type in his number before call forwarded) - Remote Agents get forwared call on SIP web/desktop software that shows member info - Agents Report call to member with attached link to re...
We need to finalize configuration of IVR in freeswitch install based on detailed workflow (see file) user inputs some general settings and local settings and has his IVR in each DID activated according to his settings use TTS system to generate member prompts (for example for welcome to my company in various languages (google TTS) System language default prompts depending on DID or member member uploaded prompts for various parts of the IVR standard workflow Incoming calls routing depending on member ivr setup (voice mail / agent / sip / skype ) Detect callerID member can set call to go to agent (according to rules and availability etc.) or to SIP or to Skype or to Gtalk
...is familiar in working with FreeSwitch. we would like to create an Admin and Customer Portal that uses FreeSwitch to provision DIDs, set SIP, Origination, Termination, Hosted PBX, Attached is some screen shots of how we would like the Customer Portal to look. We would also like the capability of using Open Source Application Development with Voice, SMS, IM and Video There is an open source Application Platform called that we would like to use their API on our FreeSwitch also to allow customers to create Voice, SMS, IM and VIdeo Applications. You must have recent experience with FreeSwitch or we can not move forward. If you have experience using Asterisk, A2billing and have customized VOIP Portals this will be ok, but you must know about FreeSwitch...
I provide you with an SBC (ie:) I connect this SBC to a gsm modem (ie: huawei e220) with usb cable. The gsm modem receives incoming MMS. Those MMS need to get picked up by the SBC and the SBC must httt post them to a url. The job is to build the application served by the SBC that 1) picks up the mms from modem 2) post it to a url This must be professional grade application able to deal with the load. Both the sbc and the modem will be operated from a remote location. Thus : the application must be able to reset/restart the sbc and the modem. The application must be able to erase phone numbers as needed (security reasons).
I need freeswitch support for 6 hours to set up a wholesale route.
We need help creating a 60cm LVDS cable between LCD panel - G121X1-L03- or G133l1-L02 (enclosed datasheets) to the following SBC -T3730 (enclosed link) we need 5 cables, thin as possible without any additional boards - Direct LVDS cable to the 30Pin LCD panel. Let me know if you have any question.
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I am looking for a service/company doing device or hardware hosting, in India, Russia and Turkey. This is hosting of my gsm modem (ie: ) which is connected to my SBC (ie: ). Note that my gsm modem must be close to the cellular signal (ie : window) to receive sms. The result of this job is a complete professional grade hosting solution found for each of the 3 countries.
Phase 1: Set up a virtual PBX (Asterisk, FreeSwitch, e.g) with a couple of key features: * Record the voice mail recording to .wav or .mp3 (.ogg) of the phone number called, but eliminate the rings as much as possible. That is, keep the call compressed to the speech portion of the VM recording without cutting any voice. * Determine who the Local telecom provider is of the phone number being called by parsing the call setup and billing string. Please outline how you would go about this function. This data will be stored in a database along with datestamp, phone#, Local provider, and sound file. * Flag for busy, endless ring, or reorder signal I do not necessarily need a gui. I am very comfortable on the command line in *nix. I am open to using a third party P...
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.
Seeking high-quality, responsive/agile software shop which has the following skillset: Software: Embedded software development in Linux, using C/C++; as well as in interfaces (I2C/SPI and similar, e.g. USB, RS232…). The areas of work would include 1) Developing a software package for a single board computer (SBC) with tuner control and narrowband FM demodulation, and 2) Creating a GUI/display and communication-with-SBC program on a PDA A) Could you please confirm to what extent your firm is a match with the above? If there are some areas which you do not cover (for example, if you only provide #1 but not #2), that can still work. In such case, please let us know whether you have sources that you recommend for those areas. Alternately, if you would have rec...
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
Write software in C language for Data Acquisition Units. The hardware is BL4S100 SBC. • Setup 15 GPIO to detect changes in its input. • Setup 5 Analog input to periodically scan for a pre-determine value. • SetupWeb-sever to display status of GPIO pins. TCP/IP stack already done. • Setup 2 serial ports. One for interfacing with a wireless modem and the other to interface with PC/Keypad. • Information/packet form PC/Keypad are handle by performing of the 14 specify task. These 14 actions will be detail later. • Information/packet from wireless modem are also handled by conducting any of 6 specific actions. Actions are to be specify later. • Any changes in the 15 GPIO pins are handle by performing a total of 3 actions which are to be sp...
Hi guys! We need to create a very simple joomla virtuemart web shop that looks like this: 3Bcat104795280% 3Bcat104779080 It should not be charged content, we will own. Customers want a simple web shop. For details, we will send ... Good luck
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: <>
...review PLease submit relevant examples of quantitative and qualitative work product. Active Owners Fund Master Template Instructions 1. OP Model Tab - Spread numbers of target company (i.e. historical and projected revenue by segment, revenue drivers / metrics, last 3 years and last 8 quarters, from revenue down to Net Income ) - EBITDA calculation - add back non-cash charges (i.e. SBC expense), 1x charges (i.e. restructuring, compensation, insurance, etc..) and show pro forma EBITDA - Also show PF EBITDA - Capex - Show y-o-y change in revs by segment as well as q-o-q sequential changes - Show Margin analysis - gross margin, op margin, EBITDA margin, Net Income margin, etc... - Show Op expense ratios - SG&A as % of sales, R&D as % of sales, etc... FOO...
...AutoCad and lacks a creative and professional appearance. Existing Logo shows the existing text based logo of the company. Tasks: 1. Re-create the small house "cabana" so it doesn't look so mechanical, utilize the same detail but make it look more natural and artistic with a simple blended tropical color theme. 2. The SBC typography should be used again but needs to interact with the cabana. Conceive a way to incorporate the illustration and the SBC together. Logo will be used for corporate identity, apparel etc. 3. "Construction" can be the bottom horizontal base of the logo while the image and text mesh together above. 4. Create several versions according to the tasks outlined 1-3. Color schemes will be revised during...
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...
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I'm looking for someone with expertise who can modify existing Freeswitch endpoint modules and can create custom endpoint modules from scratch.
...an effective SBC. The solution would need to be able to be set up on a vmware virtual machine running centos 5.5. OpenSBC or similar can be used, but we would need a proper documented project. The SBC should provide as minimum the following: - Effective topology hiding of internal asterisk/voip network - Multiple NICs as needed - Ability to provide HA failover to another machine running the SBC - NAT traversal for devices behind far end residential or office firewalls (ie no remote firewall configuration required) - Effective defence against denial attacks and spoofing - Provide an effective SBC (as if the device was not in place) so that functions such as call control, transfers, BLF, MWI, presence/status are completely unchanged. - Configurable ...
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...
