Kamailio freeswitch fusionpbx jobs

We need deploy a 8 server setup of Kazoo, distributing services as recommended by 2600Hz, the deployment includes 2 kamailio servers - kazoo-UI, 2 freeswitch servers and 4 big couch servers. 3 Default Trunking providers will be setup, none of them use the default dialing plan, so that must be customized according to how each carrier process the dial plan. DID formatting must be customized as well. (Non US) Billing must be working within Kazoo and specific rates for each customer would be required. More information about the product to deploy can be found here:

I need to know requirements . System operating . Hardware / Software . some training about use freeswitch api for .net application So I can start a server .

I have a working FreeSwitch installation on CENTOS. I want a new Ubuntu server configured so I can move my XML files on to it. No bid over $100 will be considered.

Hi, I am looking for someone who can setup a stable and easy manageable SIP server. We get call from a SIP trunk and we want to be able to also make and receive call from other trunks. It must be easy for me to create extra users/extensions, groups, DID and setup trunks. I was thinking about OpenSer/Kamailio with a webinterface. Can you help? Erwin

We are a VoIP Carrier and Voip provider based start-up company looking forward for fresh Installation and configuration of Freeswitch with Session Border Controller and Asterisk with OpenVPN with various trans-coding protocol to use based on Least Cost Route for Inbound and outbound Traffic. Telephony/VoIP/IP protocols like :SS7, SIP, SIP-I/SIP-T, H323, ENUM, RTP, RTCP, ISDN, IPSec. VoIP Configuration and Programming FreeSwitch Installation & Configuration SBC and OpenSip Asterisk and OpenVPN for Transcoding and Routing Server with Codecs to use constrict bandwidth G.723 and G.729 etc. Auto Provisioning of SiP devices Billing Management ( A2Billing) Customization DialPlan Setup Interface CallShop Setup with Bit of Programming required to customize UI (Python/P...

We are a VoIP Carrier and Voip provider based start-up company looking forward for fresh Installation and configuration of Freeswitch with Session Border Controller and Asterisk with OpenVPN with various trans-coding protocol to use based on Least Cost Route for Inbound and outbound Traffic. Telephony/VoIP/IP protocols like :SS7, SIP, SIP-I/SIP-T, H323, ENUM, RTP, RTCP, ISDN, IPSec. VoIP Configuration and Programming FreeSwitch Installation & Configuration SBC and OpenSip Asterisk and OpenVPN for Transcoding and Routing Server with Codecs to use constrict bandwidth G.723 and G.729 etc. Auto Provisioning of SiP devices Billing Management ( A2Billing) Customization DialPlan Setup Interface CallShop Setup with Bit of Programming required to customize UI (Python/Php) Sip D...

I need a native app. It should target all mobile devices. I would like it designed and coded. I'm looking for freelancers only .. you should be very familiar with freeswitch. and rock solid command over VOIP security

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 i...

You profile: You need to know enough freeswitch. Languages possible to use for web UI: 1. Priority is Haskell (possible to use Frameworks) (without experimental libraries) 2. C/C++ (possible to use framework WT) You are able to modify some code in freeswitch packages in C/C++. You need to know well nibble bill mod. You need to be able to make connection between PostgreSQL databases and clustering. Task. Before you accept project you will receive file with instructions and if you can do that then you can accept project. 1. You need to code our "sketch" algorythms and code only 2 USER's area web GUI and just 10 variables for Admin WEB UI. You NEED NOT to develop the design for that it must be just simple view with black and white webpage. May be you need t...

Hi, need to make a script for freeswitch in chatplan make a bridge if receive a sms from user and use the number inside bode of sms

Hi, need to make a script for freeswitch in chatplan make a bridge if receive a sms from user and use the number inside bode of sms

You profile: You need to know enough freeswitch. Languages possible to use for webGUI: 1. Priority is Haskell (possible to use Frameworks) (without experimental libraries) C/C++ (possible to use but if you have a lot of reason that you can prove to us) Python (without frameworks!) Perhaps you need to modify some code in freeswitch packages in C/C++. You need to know well nibble bill mod. You need to be able to make connection between PostgreSQL databases and clustering. Task. 1. You need to code our "sketch" algorythms and code only USERs area webGUI for that with some extra functions it is mostly work with nibble bill that freeswitch has. May be you need to correct our "sketch" algorythms. 2. You need to write paypal/skrill paymentgateway o...

Title: Dsocument how to setup Kamailio (SIP server) & FreeSwitch (multi-protocol softswitch, supporting SIP) backed by Cassandra database I am creating a mobile application that uses VOIP and IM, but I need assistance documenting how to setup my server. Proposed architecture: Voip Phone (SIP client) -> -> software load-balancer -> Kamailio -> software load-balancer? -> FreeSwitch (only in UDP) -> Real VOIP Provider. Kamailio - use Cassandra backend database - DDOS detection (WITH_ANTIFLOOD) - use with FreeSwitch (WITH_FREESWITCH) - TLS only FreeSwitch - Only communicate with Kamailio over UDP - Can there be multiple FreeSwitch servers for each Kamailio? Is it necessary during hi...

I am looking to add basic Voxbone integration into Kazoo. Kazoo is a voip platform with is redundant and very scalable. It is based on erlang and uses freeswitch and kamailio, along with bigcouch (couchdb ). Already Kazoo has support for Vitelity API for the same uses that I need DID purchase via a vitelity provider module. Kazoo was designed to have these modules developed and used with the platform. I need to integrate Voxbone API so that users are able to automatically buy DIDs on the platform. This involves creatring a module in erlang just like the vitelity module. Links to the vitelity module:

Dear Freelancers, We are looking for someone who can build optimization in voip traffic. Please find the below key points. 1. We have Freeswitch and softphone clients in android and IOS platform. Android Client source is csipsimple and IOS is Siphon. we need make optimization between switch and client. Actually we using G729 codec in switch and it is consuming around 48 Kbps for each call Because of overhead. Please refer a product that currently available in voip business " Itel Byte Saver" you will get more idea after reading about itel byte saver' We need slimier like that or same :) We need implement below technique to get better result. 1. RTP Multiplexing and Delta- Multiplexing 2. Voice / Packet Header Compression 3. Packet Header Reduct...

You need to know enought freeswitch. You need to be able to write python without frameworks (pure, checked python without expiremntal packages). Perhaps you need to modify some code in freeswitch packages in C/C++. You need to know well nibble bill mod. You need to be able to make connection between PostgreSQL databases. Task. 1. You need to write in python our "sketch" algorythms and make webgui for that with some extra functions it is mostly work with nibble bill nad freeswitch. May be you need to correct our "sketch" algorythms. 2. You need to write paypal/skrill paymentgateway or you can use some ready paymentgateways to modify it for us. 3. Finally it must be archive with all skripts and configuration files for Centos 6.6. You can not ...

...service is similar to googlevoice, but users can use their browsers to make calls. I am currently using freeswitch as my PBX. I want to use a webservice api to control my freeswitch instance. I need a service provider to create 'hooks' into freeswitch. (Using custom modules, existing functionality, or hacks) I have three needs: Remote LDAP 1. When users login to the freeswitch server I want freeswitch to make an http post request to a webservice to see if that user is valid. The request will return true if the user can login. (post /login- {username:login_username, password:login_password}) Pre-call check 2. when a user attempts to make a call, I want freeswitch to send a http request to the webservice, it will re...

Need someone to build and compile to run on my nexus 10 4.4.0 Then if it works with freeswitch I will post a private project to add some features

...Arodrigue. We will need to install the following softwares in our server: - A2Billing software - full installation - Asterisk 13.1 - Mysql / MariaDB - full installation - stun-server - NTP server - DNS server - HylaFAX - full installation - AvantFAX - iaxmodem + spandsp - Apache HTPd server - Java tomcat server - e-mail server - Postfix + dovecot + spam-assasin -+ calmav - AVChat software - Freeswitch software - PostfixAdmin - SVN repository Will you be able to manage the installations of the mentioned softwares, and if yes, please inform us about conditions? The ultimate goal is to create a completely new communications platform (based on the latest versions of the mentioned softwares), following the example of the existing one we have and developed for our needs. (ple...

Hello, We will need to install the following softwares in our server: - A2Billing software - full installation - Asterisk 13.1 - Mysql / MariaDB - full installation - stun-server - NTP server - DNS server - HylaFAX - full installation - AvantFAX - iaxmodem + spandsp - Apache HTPd server - Java tomcat server - e-mail server - Postfix + dovecot + spam-assasin -+ calmav - AVChat software - Freeswitch software - PostfixAdmin - SVN repository Will you be able to handle the installation of the mentioned softwares, and if yes, please inform us about conditions, and also about support in the future? The ultimate goal is to create a completely new communications platform (based on the latest versions of the mentioned softwares), with voice calls, fax, Audio conferente, web video ch...

I need a trusted kamailio expert to setup a secure communication using zrtp. You must have experience in kamailio as i need to do some stuff on a VPS. Only ADVANCE kamailio experts please.

I am looking to integrate Multiple ASTPP+Free Switch servers with a Kamailio server or Kamailio servers. Note: The Kamailio server(s) will handle the actual call routing (incoming and out going), while ASTPP + Freeswitch combination handles the IVR, etc. But, Authentication, Call Control, Credit Control functions could be shared between the ASTPP + Freeswitch servers with the Kamailio servers.

I need FreeSwitch and vBilling on a VPS - I need a fully working installation. It will be tested after the install is complete.

Hi,, Freelancers, we have installed freeswitch on a server and configured well. But we see losing packet from server side to user side. If any one know about freeswitch and the problem please bid. Kindly please check below screenshot that captured using wireshark. Thank you

...connections with the mobile network operators to route voice traffic via SIP connections to our cloud-based application servers, and run a local IVR, in addition to local SIP phones. We are currently seeking an IP Telephony Engineer with extensive skills and experience to deploy, operate and maintain a Plivo PBX server (FreeSWITCH-based) in our Kabul office. Duties & Responsibilities: • Implement and maintain VoIP telephony server using a locally installed Plivo (FreeSWITCH-based) server. • Connect existing E1 connections (via SIP over VLAN from operator) with local Plivo server. • Connect cloud-based IVR application server (Twilio, Plivo or similar) with local Plivo server. • Support access to the Plivo server using it’s RESTAPIs or...

...year, and I would like to hire voice engineer with the following items. a. Initially only 8ch of simultaneous calls. (Design must allow growth to like 1000 ch) b. Needs to be expandable to like 1000 simultaneous calls with 100000 DIDs. c. Must have fault tolerance with kamailio redundant system. d. Both wholesale sip trunk to sip trunk connection & retail service for hosted pbx for multi-tenant required. e. Billing platform with RADIUS connected to kamailio required. f. Coordination with carriers. 1- kamailio, asterisk, realtime, load balancing 2- Billing for asterisk with prepaid, multi-tenant 3- web portal for admin, client for multi-tenant, IP PBX management, prepaid and postpaid system with calling card feature. 4- test and debug The server ...

...year, and I would like to hire voice engineer with the following items. a. Initially only 8ch of simultaneous calls. (Design must allow growth to like 1000 ch) b. Needs to be expandable to like 1000 simultaneous calls with 100000 DIDs. c. Must have fault tolerance with kamailio redundant system. d. Both wholesale sip trunk to sip trunk connection & retail service for hosted pbx for multi-tenant required. e. Billing platform with RADIUS connected to kamailio required. f. Coordination with carriers. 1- kamailio, asterisk, realtime, load balancing 2- Billing for asterisk with prepaid, multi-tenant 3- web portal for admin, client for multi-tenant, IP PBX management, prepaid and postpaid system with calling card feature. 4- test and debug The server ...

Hi, iam searching voip developer to help us to create/modify wolrd wide voip platform based on Opensips,Freesswitch, Cgrates. Right now we have working platform but we need additional pair of hands :) to help us with planned work. If You have expertise in OpenSIPs and Freeswitch we need yours advise with voip design and coding in LUA. You may contact me directly at marcinm() or skype

...click 2 call engine using freeswitch (preferable). The concept is: An end user is visiting my web site. He see's a windows that he enters his mobile phone, and clicks a call button. An automated call is sent to my (predefined in a script) phone number using my server's SIP trunk. I pick up the call, and hear a wav file playing saying : This is a click2call phone. An automated call is performed to the mobile phone that the end user has stated in my web site, using my server's SIP trunk. The system is bridging the call's so that i discuss with the end user. I prefer for this solution to use my existing Freeswitch server, instead of using Asterisk ( for security reasons). When you bid, make sure that you state, if you can configure it using ...

I need someone to send me a working config file for setting up kamailio to act as an outbound proxy for webrtc sip clients. An example javascript client to be done in jssip or sipml5. You will provide me a kamailio config setup for websocket outbound proxy.

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...

I have a kamailio server that needs a couple of fixes - adevertise correct IP - ensure voice is carried through I am behind a NAT on AWS. Its a quick task for the right person. Thanks

...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.

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 ...English 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 - ...

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.