Kamailio freeswitch fusionpbx jobs

I am using FreeSwitch for IVR alongside Jerasoft billing for calling cards. I am having a problem whereas the call is not disconnecting after the allotted call time has been reached. Need to fix this.

I need a configurable answering machine detection for FreeSwitch. Price dependant on experience and understanding of PAMD.

the default freeswitch code which uses the sofia sip library has an issue when it replies with bad from header if the client device sending the call doesnt have a properly formatted from header with the caller id name field in double quotes so i want some1 to edit the sip_name_addr_d function which parses the from, to and contact header such that if the from header has a name without double quotes then simply remove it or ignore that portion. it needs to be done for the from header only and not the other as this function parses for from, to and contact headers

Hi Dear ! i want make IVR solution, SoftSwitch, Number Management and billing platform on FreeSwitch DID numbers of diffrent countries from diffrent carriers will connected to our server all numbers INBOUD To IVR and Premium Rate Numbers

Hi Dear !! i want make IVR solution, SoftSwitch, Number Management and billing platform on FreeSwitch DID numbers of diffrent countries from diffrent carriers will connected to our server all numbers INBOUD To IVR .

I am looking for experienced person who has working knowledge on Kamailio and Freeswitch integration. I need online training for the integration of Freeswitch with Kamailio. At the end of training I must have running Kamailio with Freeswitch. Kamailio will handle IM messages. It will act as an Proxy Server. Freeswitch will be used as media server. Kamailio configuration must be able to load balance more Freeswitch Servers. Functional system is requirement of this contract.

I need a Lua script example that 1. Plays music on Leg A 2. creates a new session - orginates the call 3. on Answer wait 3 seconds 4. DTMF on Leg B 5. Bridge call 6. Stop music playing on Leg A

I have a custom built, telephony app built using Freeswitch to broadcast calls. I would like to discontinue hosting Freeswitch software and use a 3rd party or SAAS service like Plivo instead. The entire app (front end) was built in php and compiles and stitches together groups of wav files to create custom messages which are sent to recipients phones. The dialer logic and dialplans were built using javascript, XML and LUA with a C++ for VM detection. If you have experience setting up dialers and have experience in these languages, please send me your work summary and proposed solution. I want to connect my app to the Plivo API to handle all the outbound calls and retire the freeswitch server so I don't have to worry about maintenance. All the audio wav files...

I need someone to install and set up kamailio sip software on a linux server.

...developper who are working on a new website. So, we are able to do everything (front-end / design and all). We have to add webrtc capabilities to this website we are setting up. We want to have all the features that a digital PBX can give (web rtc call, video call, group call, video conference, etc...) We aren't specialize in VOIP. We need a well set up server At beginning, we were working on freeswitch. We liked it until we discover transcoding video wasn't support at the time of our RD. (the version 1.6 now does) We are opened to Asterisk too. Here is what 1) api for webrtc extension/account management (crud) 2) api for list audio call recording 3) api for list envideo call recording 4) api for conference call 5) api for call history 6) api for voice...

Hi Need the above configured and setup with g729 module loaded

I have a freeswitch softswitch that receives calls from softphones and sends the calls to a couple if Cisco gateways. The problem is that th INVTE sent to the Ciscos contains non-standard headers. I need the Freeswitch to use only standard SIP headers so the Ciscos can yerminate the call properly.

Hi I need freeswitch astpp and opensip to be installed linked and configured urgently

Hi I need a sip proxy server as a frontline for authentication to my voip server ,mainly due to security be familiar with asterisk ,freeswitch ,astpp .a2billing in order to do the proxy server.

Hi I need an expert in linux and voip systems ,I need freeswitch to be installed on debian operating system and astpp.

Dear Freelancers, We are looking for a tech person who can implement a tunneling system in voip from smartphone SIP client to Freeswitch Server (which we use as a switch for sending calls). Required implementations mentioned below. 1. Implement VoIP Tunneling service Between sip client and Server ( Clients- Android and IPhone. Server- CentOS with Freeswitch) 2. Implement Packet Reduction or Header Compression in SIP client ( Android and Iphone) android we using Csip and iphone siphon source codes. we will provide current app source code for implementation. Please bid only if you know and understand above requirements.

We have FreeSWTICH installed on our Ubuntu server. Now we need to configure it with webRTC client "". Work ground would be to configure freeSWITCH for webRTC with Sipml5, users of freeSWTICH can log in on this website and will be able to have Audio communication with each other.

Hi, I want to create a Salesforce CTI connector which will integrate with FreeSwitch. I need someone to work on a project to implement the required functionality to FreeSwitch to be able to communicate with SalesForce Open CTI connector (which will be build by another party). Thanks, Ang

...contact and further discussion. I would like to discuss the phases and asses the costs involved in completing each phase. As i have a technical background in Voice/SIP technologies i would like to work with the right person/company who has experience or a strong understanding of voice protocols, webRTC, HTML5, SIP, TAPI and various PBX technologies such as CISCO UCCE, UCCX, Avaya, Asterisk, FreeSwitch, Broadsoft. I want to be able to work with a team where collaboration can occur. Strong english communication and comprehension is a must and i am looking for enthusiastic staff to work with me on the project. Creativity and Innovation qualities will be looked at favorably. The project requires understanding of the following skillets: Open CTI JavaScript Visualforc...

I need somebody who is experienced with Freeswitch in order to help me setup our voip infrastructure.

I am looking for someone to work with me long term on a part time bases. you need to be pretty advance is asterisk and kamailio. you will be working under another another team member who has built the telecom system. i am looking to pay $300 a month on a constant part time bases. if you think you know what you are doing then please get in contact, must speak English on skype, many thanks

We require Kamailio and Freeswitch Real Time integration. Implementation and Architecture should be well documented for us to manage the system. Architecture should include 1. N + 1 instance of Kamailio - Responsible for users registrations and user to user audio/video calls 2. N + 1 instance of Freeswitch server - responsible for voicemail, conferencing, pstn and media 3. Cgrates on Freeswitch for billing 4. Geo-Clustered MySQL DB for Kamailio/Freeswitch 5. DNS/Latency routing for Kamailio FE Please indicate reference jobs of this nature. Also skype to understand exact requirement

Hi interested to install freeswitch with gui who manage usa cc traffic with lrn please contact me to talk more about the project

Hi interested to install freeswitch with gui who manage usa cc traffic with lrn please contact me to talk more about the project

i need to install and cnfigure freeswitch with GSMOPEN module, to allow make calls while connected to internet. and have full control over how calls will be handled( max duration perday, delay between calls, maximum unique numbers per day or month ...) . and a a web interface to manage sim cards (topup, check credit...). also a script to donwload content of youtube, fb ... over 3g from simcards. using vwdial or something other. also cards need to allow recieving calls and sending sms ussd automatically, based on the parameters we set. also need to detect blocked cards if they are blocked. and stop them.

I need somebody who is experienced with Freeswitch in order to help me setup our voip.

Looking for C++ programmer to develop custom .so module for Kamailio. Project duration - 4 month. Required monthly support @ around $1500/month

I want to enable sip calls directly from a web browser (Chrome, Firefox, IE) without installing any plug-ins. I am looking for someone to build a WebRTC proxy (with java SIP libraries) to sit between the web browser and my Kamailio SBC (or asterisk). I want to enable voice and video calls as well as instant messaging (browser to browser and browser to PSTN). I would like to have a step by step manual produced how to install and how to configure the proxy (preferably on Debian Wheezy).

Hi, we have a new freeswitch server that is working fine with some SIP clients and one trunk working succesfully. We can register an android phone and softphones, but for any reason, we can't register none of the both Linksys PAP2 devices that we have. This is not the famous G729 issue, also tested using just PCMU and PCMA (ulaw /alaw codecs) but nothing. Also, those PAP2 devices work fine with other external SIP accounts of other providers. The log in freeswitch say this: 2015-03-23 18:28:03.842108 [WARNING] sofia_reg.c:1742 SIP auth challenge (REGISTER) on sofia profile 'default' for [1000@] from ip And nothing else, the PAP2 show this in the main info page: "Can't connect to login server" Also, this PAP2 device can connect...

Hi, we have a new freeswitch server that is working fine with some SIP clients and one trunk working succesfully. We can register an android phone and softphones, but for any reason, we can't register none of the both Linksys PAP2 devices that we have. This is not the famous G729 issue, also tested using just PCMU and PCMA (ulaw /alaw codecs) but nothing. Also, those PAP2 devices work fine with other external SIP accounts of other providers. The log in freeswitch say this: 2015-03-23 18:28:03.842108 [WARNING] sofia_reg.c:1742 SIP auth challenge (REGISTER) on sofia profile 'default' for [1000@] from ip And nothing else, the PAP2 show this in the main info page: "Can't connect to login server" Also, this PAP2 device can connect...

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I have installed freeswitch on my raspberry pi v2 audio from the mic in on the pulseaudio device sounds great however output audio sounds like chipmunks have taken over my box. I will work and give ssh access to the box but I need the audio working correctly

Set up Opensips/Kamailio platform. Program VoIP client for the platform, which will register and dial into conference automatically (get meeting ID from Cisco WebEx API/SDK). take look at TSP Server/TSP Adapter part in Cisco develop Zone. the document is here: Providing and client source code to Jowin. beside the one time payment, we also provide annual maintenance payment to you 1000+USD per year.

...people who have experience with ASTPP and FusionPBX ** We need to implement an ASTPP billing box being a trunk for fusionpbx, later we will connect more fusionPBX servers. == ASTPP - I will provide you an ASTPP pre-installed system - You will install open G729 and apply configuration in this ASTPP server - If possible, you will install fail2ban or what you think is better to secure the server. - You will provide info (or just links) with recommended security for this server. - You will provide info about how to manage the integration with other PBX servers - We will test calls locally using G729 protocol to see al works fine. == FusionPBX - I will provide a preinstalled FusionPBX - You will install and apply open G729 in this ...

We have a new installation of NewFies Auto-dialer and we need someone who can troubleshoot why its not making calling could easily be a problem with FreeSwitch or something simple... We have no experience with this. If you know what you are doing and can help this could easily turn into a log term thing.

I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. Full features will be like the following products or I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understanding of all Codecs.

...Keypad to normal black one -Change About -Remove in contact option *all* only Sip contact available Customized setup assistant with 2 options -Make new IronCall account (No email needed,) -i Already have a Ironcall account Standard settings: -ZRTP/SRTP Enabled (ALWAYS) -TLS/SSL Enabled (ALWAYS) ------------------------------------------------------------------------------ Freeswitch Server: Account setup with IronCall App -Registration from the Ironcall app without email Server configuration: -ZRTP/SRTP Enabled (ALWAYS) -TLS/SSL Enabled (ALWAYS) -Only Internal call (extension to extension) -Sending media files enabled -Video calls via ZRTP enabled -Install SSL Certificate (We supply one) -Forward secrecy (SSL) -------------------------...

We're looking for a remote Python developer to work with us on our Open Source project called FreePy (). Experience with FreeSWITCH, SIP, RTP, or XMPP a major plus. Experience with Distributed and/or Actor based systems a plus.

We are a web agency based in Europe and we are working on multiple web, mobile, ecommerce, e-marketing custom advanced projects in B2C and B2B for which we are looking for top quality experts and freelance partners. We are looking for a VOIP EXPERT with core expertise in Freeswitch or twilio for VOIp part and php or boostrap for user panels Work is for a web based telephony application that requires voip config and advanced user panels config. We supply all panels design, dedicated server, Db. We can split work between VOIP telephony and user panels or we can use existing admin panels Do not bid if not truly experienced in web based telephony applications. Serious references required. Core features of the voip application : User will be able to subscribe to a service ...

We're looking for a remote Python developer to work with us on our Open Source project called FreePy (). Experience with FreeSWITCH, SIP, RTP, or XMPP a major plus. Experience with Distributed and/or Actor based systems a plus.

I am looking for full time FreeSWITCH / Adhearsion developer I require: - good skills with FreeSWITCH development - good skills with Adhearsion - good skills in FreeSWITCH administration It would be good to have skills in: - CoreOS - - Debian / Ubuntu administration - Git knowledge

We need a product that allows us to place calls on our VoIP system from a Windows TAPI-compliant dialer and receive screen pops as well with incoming call information. Call duration should also be supported so that our CRM can automatically log the call. Our VoIP system uses Freeswitch/Kamailio and we use Polycom IP phones. We tried SIPTAPI but it does not have all of the features that we need. I would recommend using their open source version as a starting point/base. See: Need TAPI driver to work on Windows 7 64bit at a minimum. Preferably Windows 8 64bit too.

We need a product that allows us to place calls on our VoIP system from a Windows TAPI-compliant dialer and receive screen pops as well with incoming call information. Call duration should also be supported so that our CRM can automatically log the call. Our VoIP system uses Freeswitch/Kamailio and we use Polycom IP phones. We tried SIPTAPI but it does not have all of the features that we need. I would recommend using their open source version as a starting point/base. See:

Need to install a script on our server. Requirements are: 1 Setup and Run Freeswitch 2 Setup and Run Plivo 3 ICTDialer Software Installation 4. Configure 5. Configure Following open source softwares are required for ICTDialer to function fully: * CentOS 6 * Apache 2 * MySQL 5 * PHP 5.3.3 * php-mysql * php-gd * php-curl * php-imap * php-dom * php-mbstring * perl * perl-DBD-mysql * libtiff * mysql-devel * git * sox

Hello i have a Open Simulator server running on the OSGrid and i want to setup FreeSwitch Voice Server on my Open Simulator regions but ive tried everything and failed, i need you to install FreeSwitch set it up for Open Simulator and change the config in the for the Freeswitch settings and get voice working for me, you will have to TeamView into my computer and do this for me - Thanks.

...- ring sound (play in a cycle) - incoming call sound (play in a cycle) - call ending sound (play in a cycle) Find the sounds in the attachment (attachment will be provided for a chosen person). 2. Compile and send the updated file. _________________________ Flash application sources are here: There will be in the attachment. It includes an application for a local testing of flash calls. Flash mode is installed there forcibly for all browsers. Instruction for the application: 1. Open the page in the first browser tab - User Flasher 1. 2. Open the page the second browser tab - User Flasher 2. 3. If the flash application connects

Need support to assist with freeswitch installed by astpp with dialing and receiving calls and a few other things

We would like to create a new mobile VoIP app for iOS & Android for voice & text. The app will be data efficient with the potential of further development in future i.e. video call, gaming etc. The app will have pay-in features for prepayment including in app purchase. Experience in open source platforms i.e. Asterisk, Freeswitch is considered advantageous.

I am looking for someone who can help in connecting my FreeSWITCH to Avaya CM, currently I am able to register my FreeSWITCH connection to Avaya CM but when i try to make calls I am getting Error 404.. Both Avaya CM and FreeSWITCH are on same network and my softphone installed on FreeSWITCH machine can easily make and receive calls.. Looking for someone who can help with it.

Hello currently im running SoaS (Sim on a Stick) its opensimulator but with a lamp stack for windows im trying to get phpmyadmin working as there is only mysql and no web interface for mysql so ive tried that and when i go to my localhost nothing is coming up its just an empty page i need someone to fix this problem, also i want voice enabled in my opensimulator i want you to configure freeswitch on my desktop so i can talk in opensimulator with my friends who connect to it aswell