Kamailio freeswitch fusionpbx työt

...TinderPlus and Snapchat. rn*Must - Only experienced developers need apply with coding for TinderPlus and/or recommended coding for this projectrn*Required Integration - Fb, Tweeter, Linkedin, Instagram, Tumblr & Flickr rn*Language - PHP, Java, Objective-C, Ajax, C++, C#...rn*Platform & Systems - Lamp, Lemp, Android SDK, Xcode, Linux, MySql, Web API 2, IIserver, SQL server, Open talk API's, Freeswitch, Objective C, Node.js, Phone Gap... rn*Tools - Micromedia, Dreamweaver, PHP edit, Android Studio, Eclipse, Bootstrap, Asp.net MVC... rn*UI - HTML5, CSS3, iquery, Javascript, rn*Videostreaming - Soundcloud, Shoutcast,WebRTC rn*Other - ART, Titanium, Rhodes, Eachscape, rn*Contractor will pay for server and API'srn*Must Features - .Wifi enabled...

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

Install and configure FREESWITCH on CentOS so that it successfully integrates with sipml5 based web client to handle outbound calls via multiple trunks (wholesale VOIP providers)

Need to configure FREESWITCH for VOIP communication needs.

I need a FreeSwitch specialist to configure our freeswitch application correctly

...balancing and redundancy. Also to have an astpp package setup with sample customers and the ability to do monthly billing through my credit card processor at a set monthly rate. It is very similar to the document "Load Balancing of network" that is posted in your portfolio at this site. As of now I believe that I would have 1 opensips server and 3 freeswitch/fusionpbx or Kazoo servers with a common database on one of the freeswitch media server. I would have the servers setup and running CentOS 6.5 with a shell account for you to access. I was wondering if you could give me an estimate of the cost for a project of this nature. I am also curious what cost would be for ongoing periodic installation and configuration of newly added systems and s...

1-) New DID add and delete added dids function: * Example: http://ip/ * Example: http://ip/ 2-) Outgoing call script: Making calls via URL's. There are three parameters in this script. * The DID number to call will made. * TO number (this number will be called) * Soundfile name. This soundfile will be played after TO answer the call. * Example: http://ip/ (http://ip/) * If there is no soundfile named as , or no DID return an error. * Make the call with these parameters. * If user dont answer or busy, try to call again (total 2 times) * After user answer the call, play specified soundfile 2 times. * If any key pressed, post this keypress to this our API: {did}&from={tonumber}&key={pressedkey} * Hangup the call af...

I can not make the configured Elastix (3.0) MT (Asterisk + kamailio) work correctly to make and receive calls with external SIP trunks, they register, but not finalize authentication and stay with status UNREACHABLE. We need a professional who knows the asterisk and kamailio to solve problems encountered with the trunk configuration;

Sex Dating App for iOS + Android + Website *App/site name - txxxt *Domain - *Theme Q - "Wanna have sex?" *Concept - blend of TinderPlus/Snapchat using the exact same coding including some Facebook features *Suggested - Java, Objective-C, PHP, HTML5, CSS, UI/UX, ASP, NET, Django, *Recommended - WebRTC for video streaming. OpenTok APIs/ Freeswitch as a SIP server. Objective C for the native iOS application. Java for Android application. Node. JS for the server side *Design/Coloring - design similar to TinderPlus. Colours - black/purple/pink as in *Requirements – Must know script/coding for Snapchat and TinderPlus. *Must Features - In app rating system, In app purchase system, Wifi enabled calls through the app to be like a normal phone call, Vide...

Already running vTigerCRM 6.3, need to use other both tools to integrate into vTigerCRM for handling telemarketing calls of a business, vTiger handles recording for later review. Server runs in Os X Yosemite, with VMware Ubuntu server installed. for emulate linux, if needed. Its not an installation for resale, its a project running. Phone lines connected into Asterisk will be virtual from an online provider. Thanks.

What we are looking for is very simple, I guess it can be easily done with IP-PBX or Switches, what needs to be done is to setup Inbound IP Based SIP Trunks which will bridge calls with External SIP Accounts. My external SIP providers do not offer trunk, instead they are offering username/password based aut...Inbound IP Based SIP Trunks which will bridge calls with External SIP Accounts. My external SIP providers do not offer trunk, instead they are offering username/password based authentications, and they requires the accounts to be registered in order to make calls, so calls without registration do not works. Someone else has created a similar service like this for me before with FREESwitch but as freeswitch needs 729 codec to be purchased, it didn't work tho...

I have ongoing work related to our previous project 'FreeSWITCH management and configuration'

Please OK this payment for the FreeSWITCH management :) I made it to be 250 USD, could you check on the FreeSWITCH that you can add the 3 prefixes for Siptraffic route? I've added the IP now to their managementsystem. 114/115/116 = net2phone, 117/118/119 for this one?. If you need more than 25 for this last bit let me know and I'll up the price further and pay :)

...many listened to the steam) We would like to pic from phone numbers in the location of the listeners so long distance charges are not incurred. Please answer these questions with your bid: What technology do you propose should be used? Where should it / could it be hosted? What will it cost to develop program? What telephone / voip service do you recommend? What platform? Asterisk, freeswitch, etc. I don’t need to program another website as I already have a Drupal website. I just need the back-end to the platform ...

I need an advance voip expert that has a wellknown kamailio background and could implement outgoing calls to landlines and voicemail box with freeswitch or any other. Expert must solve some 407 proxy errors i have actually.

We are looking to build our own Phone system and become an ITSP (Internet Telephony Service Provider). It does not matter what the platfom is, it can be Elastix, FreePBX, FreeSwitch or Trixbox. I need someone to help building system, advise us on which cloud service to use, amazon or digital ocean or any other. I need a control panel clients can log in and manage their own phone system. I am looking for something like Kazoo or Kazoo itself. This will be on-going work, so I need someone that I can communicate with at all time and also will turn a support position to help us with our clients Desired Skills Linux System Administration, VOIP Administration, FreePBX, VOIP Software

I need an advance voip expert that has a wellknown kamailio background and could implement outgoing calls to landlines and voicemail box with freeswitch or any other. Expert must solve some 407 proxy errors i have actually.

I have some issues to fix in NewFies installation. The first is: Alert : your settings are not configured properly, please contact the administrator. To send notification to administrator Click here When I try configure the Provider in FreeSwitch, he return: Reload XML [[error near line 2728]: unexpected closing tag </gateways>]

OpenSIPs / Freeswitch / SipXecs SBC Configuration with Web GUI Administration (we are currently doing this with FreeSWITCH but we are moving past the CPS capabilities of FS) We need to create an Web GUI Admin Portal for Open SIPS - this portal needs to be able to create configs for Customer Hosted PBX that are both no-NAT and NAT capable - we trunk from our ITSP's calls to customer hosted PBX's both in the cloud and on-prem - this portal needs to be able to create a handler for INVITE's / REFERS etc. in both directions for inbound and outbound calling for the hosted PBX's Further discussion can be done if we find you are a competent candidate

We require the integration of CGRates on our existing SIP platform, a Kamailio proxy and an Asterisk farm. We need CGRate setup on Kamailio for : - Prepaid - Postpaid - Bundles (i.e. Postpaid + free minutes to destinations) Implementation and Architecture should be well documented for us to manage the system. Please indicate reference jobs of this nature. Also skype to understand exact requirement.

We have a project to install Kazoo and freeswitch in a multi tenant environment and we need help with that.

...website to that my VoIP users can sign up and be able to edit their accounts by their own. It must be integrated to Kamailio database on my VoIP server. Today, to add new users, I have to login on my server via SSH and type a Kamailio command: kamctl add <user> <password> To DELETE user: kamctl rm <user> To Show users in database: kamctl db show subscriber To Show users online: kamctl online So I need to integrate the Kamailio database with the form on my website to allow, for example, users to sign up, edit their passwords, check payment status, etc. I have found on StackOverflow a tip of how to do it: I will also need to integrate it with Stipe subscriptions

Need developer to make auto dial module for FusionPBX (Freeswitch GUI).

We are looking for an experienced administrator for our Freeswitch PBX.

Set up Siremis/Kamailio server as an SBC for routing, initially ougoing calls using LRN dip for carrier route selection.

...extract data with linux command. Then, I need to be able to search those number very quickly from the DB so you will need to do the query and the indexes setting. Then, from the numbers collect, you need to make a multi-threaded concurrent socket calls to send out those number on a 3000 call per second basis. This program is to run in Linux. Desired Skills PHP, Python, PostgreSQL Programming, Freeswitch...

Sex Dating App for iOS + Android + Website *App/site name - txxxt *Domain - *Theme Q - "Wanna have sex?" *Concept - blend of TinderPlus/Snapchat using the exact same coding including some Facebook features *Suggested - Java, Objective-C, PHP, HTML5, CSS, UI/UX, ASP, NET, Django, *Recommended - WebRTC for video streaming. OpenTok APIs/ Freeswitch as a SIP server. Objective C for the native iOS application. Java for Android application. Node. JS for the server side *Design/Coloring - design similar to TinderPlus. Colours - black/purple/pink as in *Requirements – Must know script/coding for Snapchat and TinderPlus. *Must Features - In app rating system, In app purchase system, Wifi enabled calls through the app to be like a normal phone call, Vide...

Sex Dating App for iOS + Android + Website *App/site name - txxxt *Domain - *Theme Q - "Wanna have sex?" *Concept - blend of TinderPlus/Snapchat using the exact same coding including some Facebook features *Suggested - Java, Objective-C, PHP, HTML5, CSS, UI/UX, ASP, NET, Django, *Recommended - WebRTC for video streaming. OpenTok APIs/ Freeswitch as a SIP server. Objective C for the native iOS application. Java for Android application. Node. JS for the server side *Design/Coloring - design similar to TinderPlus. Colours - black/purple/pink as in *Requirements – Must know script/coding for Snapchat and TinderPlus. *Must Features - In app rating system, In app purchase system, Wifi enabled calls through the app to be like a normal phone call, Vide...

I would like to produce an ip pbx with predictive dialer module which should based on Freeswitch. The GUI must be cross-platform, even mobile platforms. QT with X language in my mind but it is not strict. My goal is making a vicidial like application based on freeswitch which should be useable by call centers with predictive dialer module and useable by every company as ip pbx with its core package. Cordial Regards,

Sex Dating App for iOS + Android + Website *App/site name - txxxt *Domain - *Theme Q - "Wanna have sex?" *Concept - blend of TinderPlus/Snapchat using the exact same coding including some Facebook features *Suggested - Java, Objective-C, PHP, HTML5, CSS, UI/UX, ASP, NET, Django, *Recommended - WebRTC for video streaming. OpenTok APIs/ Freeswitch as a SIP server. Objective C for the native iOS application. Java for Android application. Node. JS for the server side *Design/Coloring - design similar to Tinder. Colours - black/purple/pink as in *Requirements – Must know script/coding for Snapchat and TinderPlus. *Must Features - In app rating system, In app purchase system, Wifi enabled calls through the app to be like a normal phone call, Video chat u...

We have an SSL which needs to be installed with the Intermediate Cert from Godaddy onto our FreeSWITCH Servers From here we need to then create a SIP Bridge to Office 365 Exchange UM so that we can setup our integration for all voicemail with UM in the cloud This process requires TLS negotiation and currently we are getting this error when negotiating with Office 365 Exchange UM tport_tls.c:1008 tls_connect() tls_connect(0x7f985007e090): TLS setup failed (error:00000005:lib(0):func(0):DH lib)

Sex Dating App for Apple and Android + Website App/site name - txxxt Domain - Theme Q - "Wanna have sex?" Concept - a blend of TinderPlus/Snapchat using the exact same programming. Including some features of Facebook Recommended Coding - Java, Objective-C, PHP, HTML5, CSS, UI/UX, ASP, NET, Django Suggested Coding – WebRTC for video streaming. Freeswitch as a SIP server. Objective C for the native iOS application. Java for Android application. Node. JS for the server side. Color scheme - black/purple/pink as in Requirements - In app rating system, In app purchase system, Wifi enabled calls through the app to be like a normal phone call, Video chat using Wifi, similar to facetime, skype, facebook video chat, Bing or google mapping User agreement – 18+ Payment G...

Looking for some Screen Pops from odoo CRM when customer calls in using free-switch connector and click2call function. In short, I want odoo to watch for calls coming in through the free-switch connector, and do a screen pop like vicidial or something similar. Agent will be able to document call using Odoo CRM module and and have relevant customer information. I want a proposal for how much it will cost for you to complete the module as well as an estimated timeline. One price, you finish the job. We of course would want to be able to test the module and ensure that it worked the way we expected, but if all goes well, there are other projects needing to be completed using odoo.

Looking for some Screen Pops from odoo CRM when customer calls in using free-switch connector and click2call function. In short, I want odoo to watch for calls coming in through the free-switch connector, and do a screen pop like vicidial or something similar. Agent will be able to document call using Odoo CRM module and and have relevant customer information. I want a proposal for how much it will cost for you to complete the module as well as an estimated timeline. One price, you finish the job. We of course would want to be able to test the module and ensure that it worked the way we expected, but if all goes well, there are other projects needing to be completed using odoo.

We need advice from a person with experience in Kamailio deployment (including billing solutions / Failover) in order to set up our own SIP platform which will handle calls from our carrier grade Voip network to the PSTN via an SS7 gateway (already setup). Have a look at the diagram attached. Please only apply if you have experience with such deployments, thank you.

I currently run a few Asterisk servers all sharing the same realtime Mysql database. I have users who connect in with IP authentication, and al...currently run a few Asterisk servers all sharing the same realtime Mysql database. I have users who connect in with IP authentication, and also with SIP credentials. There are problems with the current setup- A user on server 1 cannot reach a user on server 2. There is no load balancing / failover. I am looking to correct these problems by setting up 2 Kamailio SBCs which will distribute calls to my Asterisk servers. Kamailio will get all authentication information from the Mysql Realtime database either with a View or by modifying the current SIPUSERS table. It should be very straightforward for anyone familiar with th...

...should refresh automatically in (let's say) 30 seconds. Multi-Tenancy: This portal should be multi-tenant so that we can provide it to several clients through our Free Switch Server. For a multi-tenant portal you can use fusion pbx as it already has a multi-tenant system (based of free switch). We are however open to use Asterix portal if it has multi tenancy. Desired Skills PHP, Python, Freeswitch, VOIP Software, Asterix...

Setup freeswitch auto_nat in amazonaw ec2 in linux server 1) my android phone SIP application can register a SIP account to freeswitch via T-mobile 4G network and public wifi such as Stuckbuck coffee shop. sip:1@ 2) my ubuntu SIP application can register a SIP account to freeswitch via WiFi network ( home wifi router) such as sip:2@ As the result, I would see Sip 1 and 2 can call each other with video and audio.  link .  I belive the job scope is to configure sip profile with deep understanding and about UPnP or NAT-PMP and would be straight forward for hands-on freeswitchers who has done it before.

Necesito conectar una planta Asterisk a una Planta Avaya por medio de una troncal H323. En este momento tenemos problemas de señalizacion y se pueden generar llamadas via Asterisk -> Avaya pero no vie Avaya -> Asterisk. Si tiene conocimientos en el manejo de otra PBX como Freeswitch o Yate, se puede usar esta tecnologisa y subirlo como Gateway.

...(Sydney, Singapore, Seattle, Ireland) with domestic trunks to offices in Australia, Philippines/Indonesia, Canada/USA and UK/Europe. Approximately 100x hardphones (Cisco SPA 5xx) plus 50x softphones. This is not a simple/small setup but it is not massive either. Some complexities with latencies and trunk codecs are expected. You will provide:- - advice on which Asterisk frontend, Elastix, FreeSwitch etc - setup of the EC2 instances and the configuration of the Asterisk front end - assist with sourcing appropriate SIP trunk providers in each domestic location - setup of calling plans, least cost path routing, inter PBX call routing etc - setup of call queues, ring groups and other calling features as required You will have:- - proven experiance in handling t...

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.

Tenemos una plataforma desarrollada en freeswitch que ofrece servicios de audioconferncia. Buscamos personas que nos apoyen en el mantenimiento de dicha plataforma.

We need to configure 2 voip carriers and upload the rates in correct format with our profitt at our Freeswitch voip.

I have created VOIP application using PJSIP library device side and Kamailio server. Need some one who know the Networking background who help me to resolve background issue.

We like to use SIP to initiate a call (Asterisk or Freeswitch) to the Viber on linux. Proper signaling would be the main issue of this project. (Signalling Viber call status to SIP status)

i need to configure free switch for voice broadcasting in window platform,i also need cdr,billing

We have a bigbluebuttom server, where we are using freeswitch. We would like to use Freeswitch as our PBX

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