Freeswitch openbtscông việc

We need to add voice and video to our chatting application, one 2 one and groups / conference with options like answering machine, recording... we are using customized version of ejabberd 14.07 we need someone expert with freeswitch/Asterisk and xmpp, you should be able to suggest, implement and support the right solution. Message me only if you have worked/implemented such solutions before and it was success!

For our project we are currently looking for VoIP Developer. Requirements: Mandatory •Full-lifecycle commercial experience in the development of VoIP platforms •Strong OpenSIPS development skills - load balancers - registration servers - HA solutions •Strong FreeSWITCH development skills - billing solutions - multitenant solutions - geolocated servers - HA solutions - scripts in LUA - anti fraud solutions •Self-starter who is able to "hit the ground running" •Familiar with structured software development / quality processes •Experienced in the creation and use of requirements documents, specifications, designs, unit test plans etc •Demonstrable communication and team working skills Preferable •VoIPmo...

...General Description: Installation of Freeswitch and ASTPP billing on Debian 7.5 Wheezy. Work expected: -Job must be documented in detail, we need to be able to replicate the installation and configuration by following a simple and descriptive list of commands and/or tasks. -After the ASTPP is fully tested using SIP Accounts, Inbound and Outbound calls from our DID's and SIP trunks the linux installation will be wiped out, one of our employees will attempt to complete the above documented work, if he or she is successful then the project is considered complete. More details: The installation will take place on a testing environment installed on a VPS. Applicant requirements: -Must be able to communicate in English -Experienced on Freeswitch and ASTPP i...

I want a turnkey solution for click2call integration using freeswitch. I will need the code for the plugin, and freeswitch configuration, on my already installed freeswtich server.

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP...* Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understanding...

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP...* Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understanding...

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP...* Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understanding...

I wants to integrate a SIP interface between Avaya and Freeswitch,

Linux Install FreeSwitch/FusionPbx + PostGreSql + Kamalio/Fail2Ban 1)Server A: Linux - Install FreeSwitch/FusionPbx - Point FreeSwitch/FusionPbx to external PostGresql database server B 2) Server B: Linux - Prepare Postgres to accept external connections 3) Server C: Linux - Install Kamalio/Fail2ban/whmc Skills required: FreeSwitch/FusioPBX, PostgreSQL, PHP, Python, Django, javaScript, C/C++, XML, mod_Lua, Call control xml, voip Note: Payment is after project testing and approval. Regards

Build FreeSwitch/FusionPBX custom Billing and Callback Script - Build custom Billing for FreeSwitch/FusionPBX - Build custom callback and Callthru script for FreeSwitch/FusionPBX Skills required: Perl, Python, javaScript, Lua, C/C++, Java, PostGres NOTE: Payment is only after project testing and approval. Regards

Configure FreeSwitch/FusionPbx x 1 server + Postgre database x1 server a)Server:FreeSwitch/FusionPbx running - Install FusioPBX on running FreeSwitch server - Point FreeSwitch server to use Postgre server - Point FusionPBX to use postgres server b)Server: Postgres database running - Prepare Postgres to be used by FreeSwitch and Fusion. - Secure database. Skills required: Perl, Python, javaScript, Lua, C/C++, Java, spydermonkey, PostGres mod_lua Note: Payment is only after project testing and approval. Regards

0. install FreeSwitch 1. Connect to SIP provider (Asterisk) 2. Setup codecs & protocols (T.38 is a must with G.711 fallback) 3. Install GeoFaxIp. 4. Connect Hylafax (installed) to GeoFaxIp. 5. Make everything work T38 / G711 Fax protocol. 6. Make Hylafax pass the CID to use (what number will the destination see) to FS to Asterisk. The Freelance should know FreeSwitch, ssh, *wireshark*, debug SIP Protocol problems, and Linux in general. The project is finished when Hylafax sends faxes using T.38 / G711 protocol via GeoFaxIp->FreeSwitch->Asterisk, and have high success rate (95%+) Time frame for installation : 7 days. Please give me a price quote for the full work.

...en servicios de comunicación de voz en mercados emergentes con un rápido crecimiento, ofreciendo a carriers calidad y alto rendimiento de la red a través de interconexiones directas. Estamos buscando a un Programador con experiencia que pueda crear aplicaciones para VOIP, conference calls, chat y streaming audio utilizando Asterisk y otras tecnologías de código abierto (por ejemplo OpenSIP, Freeswitch, SER, etc.) Responsabilidades: La persona elegida trabajará en un entorno dinámico y estimulante, desarrollando e implementando soluciones únicas y personalizadas para satisfacer las siempre cambiantes necesidades de la industria de las telecomunicaciones y a la vez aprovechar las oportunidades de mercado que se presentan. L...

Install mod_nibblebill module on FreeSwitch server. Build API to connect to dashboard on custom website. a)Install mod_nibblebill module to server:FreeSwitch/FusionPbx + Postgre database. b)Build API to connect freeSwitch/FusionPBX to dashboard on custom website. Skills required: FreeSwitch, FusionPBX, Perl, Python, javaScript, Lua, C/C++, Java, spydermonkey, PostGres mod_lua Note: Payment is only after project testing and approval.

...calls. Both the scripts are fixed with different variables. (details will be disclosed while interview) We are having a preference to use FreeSwitch open source technology to implement this facility but we are still open for any other better suggested open source solution if it can provide easy implementation and better stability. There will be many things which will required to give manageable from super admin section like upload/download/change order of common WAV file scripts, Hair Saloon admin to manage WAV file of saloon name, specialists' names, etc. We are looking for serious expert with past experiences in dealing with VOIP, Freeswitch and similar technologies. We are expect bids with some examples of past implementation and your experience of devel...

Problem We have a SIP server which does not handle multiple interfaces well. The primary interface is used to serve clients (Network "A") Another interface is needed to access an upstream provider (Network "B"). At the moment we have a work around where a Asterisk relay is in place on a separate server ( so call flows from A can be pass...6) We require a detailed listed of the ports and protocols that need to be exposed between the hosts. 7) NAT of any sort can not be used. This needs to be proxy. About us. We are confident in: The server operating system installation Networking SIP & RTP In general IT/server/networking competent However we do not have any real experience using Kamailio or Freeswitch - so help...

Hi, I am experienced in freeSWITCH, however I would like someone to write a script in LUA, which interacts with FreeSWITCH, in particular the conference module. Id like the script to interact with a backend MySQL database, and then play prompts, and collect the relevant details, those being; >Please enter your PIN >Please record your name after the tone, and listen to it again/play back/modify. And then when entering the conference, read back how many people are currently in the conference. I will provide the FreeSWITCH server, with Mod LUA running, and functional MySQL database, Id just like someone to write the rough code, so I can then manipulate accordingly. Many thanks Jon

We are looking for freelancer who is expert in FreeSwitch/Asterisk. We need to program/setup/install fully IP-PBX system in FreeSwitch/Asterisk as well as he can program/integrate the Billing and Re-seller Account also.

I am looking for someone who can set up speech recognition for freeswitch. I want a user to call in and spell their name then convert it to text. Example, someone would say "J-O-H-N" and the app would return a variable with the value "john" I currently have a running freeswitch server with the mod_pocketsphinx installed but I do not know how to configure it.

I am looking for someone who can set up speech recognition for freeswitch. I want a user to call in and spell their name then convert it to text. Example, someone would say "J-O-H-N" and the app would return a variable with the value "john" I currently have a running freeswitch server with the mod_pocketsphinx installed but I do not know how to configure it.

We have a fusionpbx softswitch set-up on our server and would like to send calls to it from an outside ip and direct the calls to another ip while filtering according to dialplan. This task is basically needs configuration skills. Will provide remote access to servers. More detailed information will be provided before starting the project.

0. install FreeSwitch 1. Connect to SIP provider (Asterisk) 2. Setup codecs & protocols (T.38 is a must with G.711 fallback) 3. Test T.38 with a fax software. The Freelance should know FreeSwitch, ssh, *wireshark*, debug SIP Protocol problems, Linux in general. The project is finished when FreeSwitch provides T.38 with G.711 fallback to client software (Hylafax) via tty or API. Time frame for installation : 7 days.

I currently use Twilio for simple PHP / REST driven telephony. However, I need to have someone setup Freeswitch / Plivo () and configure it to work with a live SIP trunk that I can provide. I will also provide root access to a CentOS server. Final outcome should be that I can make and control an outbound call from a simple PHP script (similar to how Twilio works), and the call will be made using the Freeswitch/Plivo server. Need this done ASAP (within 48 hours) Thanks!

Hi. I need Newfies/Freeswitch installed and configured on Digital Ocean with my provider which is Vitelity. I already have an install on Digital Ocean but user interface is being redone so it is not functional. You might be able to log into this one though to copy all config options to make it easy. You can probably just take a snapshot of my current install then overwrite the frontend files with the stock Newfies files. From scratch, here are docs - Ready to start now. Please ask if any questions

I want customization of Sipml5 with click2call button. I have a freeswitch/Kamailio server that I need a click2call button . We can work with other platforms like jsip, but we want most supported on browser Task 1. Customize a webrtc software eg sipml5, - 2. Provide interface for creating click2call button, we should be able to link to image 3. Click to call should anonymously initiate call to our desired extension on the SIP Server eg 1001. Website visitor should not require to login. This should also work when visiting link from mobile phone

IVR UPDATE AND MOVE TO LINUX server. PHP email fix for

I want a software Architecture and Design Documentation for Internet Telephony and Rich Messaging solution. Specifically, we are looking at a system like Viber/line, kakaoTalk. The basis of the back end system will be on Open Source software like Kamailio and Freeswitch. Also required will be a front-end application server for user management, payment sign ups etc. Finally, the system will have a rich client mobile application with minimum features of viber, line and KakaoTalk. Your Job as a Software Architect, Analyst is to produce a detailed Architecture document including wireframes for mobile app and web front end. We are not looking for trial and error bidders but rather very strong and competent Architects with skills in this area. You will make design decisions as in...

Install mod_nibblebill module on FreeSwitch server. Build API to connect to dashboard on custom website. a)Install mod_nibblebill module to server:FreeSwitch/FusionPbx + Postgre database. b)Build API to connect freeSwitch/FusionPBX to dashboard on custom website. Skills required: Perl, Python, javaScript, Lua, C/C++, Java, spydermonkey, PostGres mod_lua

Build API for VoIP Softswitch software. Build API to integrate FreeSwitch server with billing software Build API to integrate Billing software to FreeSwitch Build API to integrate with user portal on custom website. Skills required: Perl, Python, javaScript, Lua, C/C++, Java, PostGres

...a)Install mod_nibblebill module to server:FreeSwitch/FusionPbx + Postgre database. b)Build API to connect freeSwitch/FusionPBX to dashboard on custom website. Skills required: Perl, Python, javaScript, Lua, C/C++, Java, spydermonkey, PostGres mod_lua ============TASK B App uses Mysql() and is populated by mysql code. Convert Mysql code lines to Postgres code lines on app. Install and test that app connects and points app Postgre database. Skills required: Perl, Python, javaScript, Lua, C/C++, Java, PostGres ==============TASK C Build API Build API to integrate FreeSwitch server with billing software Build API to integrate Billing software to FreeSwitch Build API to integrate with user portal on custom website. Skills required: Pe...

Hi! I need SIP server running in Windows Azure cloud. I already have VM with Ubuntu but I need assit setup FreeSwitch (right now it's installed but doesn't works correctly). Let me know if you have experience with FreeSwitch and some Linux admin skills. Please only persons and only bidder with real experience.

The application is designed as a hybrid app using Cordova. JavaScript is complemented by jQuery and jQuery Mobile. The application is intended primarily for communication. The application contains a section in which you can perform telephone calls - VoIP. Telephones are connected to the server, where the PBX is running FreeSWITCH. For signaling is used on mobile websockets and as voice transmission technology is used WebRTC. It is loaded into the application as a plugin for Cordova. The call is directed phones themselves, however credentials and number to "call" are obtained from API that handles their allocation and management. The connection between iOS devices works without problems. The problem occurs on some Android devices, where there is very poor transmission of...

Hello All, I am looking for an software to be build for gathering some specific data....Do place your bid if you have good knowledge about telephony sector Need to have good track record, more detail will be given to shortlisted candidate, Reasonable price will be preferred Regards:

Hi, we have a fresh server with multi-tenant freeswitch and fusionpbx. We want to install on a separate box ASTPP and configure a set of options in it: - 3 wholesale trunks - LCR - World plans - DID management Also, users would be able to access voicemail and modify any settings like forwarding number, voice mail and other typical things. Freelancer should provide 2 things after finish: 1.- A small manual to manage all this things 2.- A monthly fee to maintain and keep this system working. Thanks,

We have a FusionPBX with freeswitch server and want to start to setup multitenants and will need to start using this install that already has G729. Also, we would like to setup some trunks and incoming - outgoing routes. If possible, we would like to setup also - LCR - Worls plans - DID management Also, in Blue Box, users would be able to access voicemail and modify any settings like forwarding number, voice mail and other typical things. Freelancer should provide 2 things after finish: 1.- A small manual to manage all this things 2.- A monthly fee to maintain and keep this system working

in a compiled opensim instance add a number of extra facilities including econmy

Install freeswitch and configure all

Install freeswitch and configure

Hello,rnI need freeswitch and Kamailio initial installation and setup training. I am a voip professional but very new to these two areas, I need your help to have a kick start of these two and then I will continue of my own.

Hello, I need freeswitch and Kamailio initial installation and setup training. I am a voip professional but very new to these two areas, I need your help to have a kick start of these two and then I will continue of my own.

I need freeswitch and Kamailio initial installation and setup training. I am a voip professional but very new to these two areas, I need only few hours of your to have kick start of these two and then I will continue of my own.

Hi Dear shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecn...shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecnical support for Freeswitch

Who can build me a GSM Interception Using USRP and OpenBTS I can buy the USRP and parts on What i want from you? - tell me wich USRP i need - tell me what addons on the USRP - tell me What PC / Labtop you need - install it fully so i can intercept phonecalls and intercept SMS - maby you can build extra's like - sms advertise let me know!

Hi Dear shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecn...shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecnical support for Freeswitch

I have a freeswitch that is database driven using node.js I need more features to it (modification to node.js) and some amount of after support. Please state your experience with fs and node.js upon your bidding

Install/configure Multitenant FreeSwitch/FusioPBX + Billing ASTPP + Database PostGre + Kamalio -Servers with CENTOS 6.5 ready for installaton x 4 a) Install all the above on Centos 6.5 Servers x 4 b) Integrate FS/Fpbx with Billing c)Configure all of the above d)Test System for going live e)Training on the system x 1 hour SKILLS: FreeSwitch/FusionPBX, ASTPP billing, PostGre, PHP, JQUERY, RUBY, PYTHON

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Installation ASTPP Open Source VoIP Billing for FreeSWITCH software on Debian 7 Linux server. ASTPP must be installed with FreeSWITCH on the same server Installation + voip test all the information here: The server is ready and comes with Debian 7 with SSH access

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP... * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understandin...

Configure FreeSwitch/fusionPBX Msql database for read/write/display from Website custom php dashboard. a))I have a Web site ready, with custom php dashboard b)I have a unconfigured FusionPBX box ready for configuring. FreeSwitch: - Extract/read/write/display user account fields/details located on FusionPBX MySQL database. Custom PHP user dashboard: - Display user account fields/details on Custom website PHP dashboard. Username/associated extension/contracted inbound virtual number/ring and pickup group/caller id number presentation option/caller waiting/follow me/Available prepaid funds sms gateway/Callback integration/ support ticket box. I need someone to: -Teach me how to implement above requirements on my FreeSwitch/FusonPBX box and Custom PHP dashb...