Hey, guys, I have a simple project. I have a FreePBX server that I need to build an inbound Sip trunk to 2 separate carriers + build inbound routes for them. It should be a simple process. Please respond to the bid with "What up Dingo" at the beginning of your message so that i know you have read.
Looking out for Telemarketing professionals with Mandarin language expertise. You will be required to work from home on our platform and dial out 70-80 calls every day and sell conference tickets. Project training will be provided
...effectiveness of their gift. Sometimes the donors want to participate with their family in the endevour but are not sure how to do that. The Legacy Giving Fund allows the investor to dial in a percentage of interest they want returned to them on an annual basis. The balance of the growth from their investment will be deposited into an account that will allow
Would like to ...to use a raspberry pi Zero to run Asterisk with chan_bluetooth module a a openvpn client a 3g usb dongle for internet Your task is to create a bootable image which I will plug into the rbpi and it will run the 3 things When device boots it will 1. connect to 3g internet network 2. connect to vpn 3. load asterisk with chan_bluetooth
...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck
I need a new website. I need you to design and build my online store. Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, busines...Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, business loan, gold loan, car loan - investments, fixed deposits, insurance, mutual fund, SIP.
The business is a provider of cloud based communications which includes: Skype For Business Microsoft Teams Cloud Contact Centre Office 365 Sip Trunking Hosted PBX We require assistance in developing and revamping our website content, feature guides, product bundles, resources for customers (How2Guides/FAQ etc) some of which we have but need updating
...underneath the hex bytes. The decimal numbers are usually rendered in most hex editor softwares as the hex bytes are selected. From here the decimal number is to be displayed upon a dial with a knob. This knob will be the addition to the hex editor by allowing the file to be modified by decimal oppose to hex bytes exclusively. Now here comes the tricky part
...forms as stated above should allow completion online and in application, there should be a live text chat at the side of each form to get assistance in completing same and to dial and engage online on a phone and or Skype. A. eCommerce payments online by phone and other should be enabled this would accept premium payments, standing orders, bill payments
Scope of work: 1. We want to integrate SMS gateway, Email integration and click to dial feature on our website 2. We want to implement a feature of live chat and chatbot on our website 3. We want a dynamic database for our website 4. We want to include Disclaimer on our website 5. We want all blogs and pages on the same URL (refer [login to view URL])
Our business is audio conferencing. We need to import minutes as line items ...this via the API to Chargify on the customers. Typically there is a minute charge for the use (eg. 200 minutes) and a surcharge for some contries on top, eg. €0.07 ekstra for dial-in to India. [login to view URL]
vicidial with c...Vici as mention will be on this widget: a. Pause/ Resume (i. Selection of Pause Caude) b. Park c. Transfer Conf d. Hangup e. Disposition (i. Selection of Disposition) f. Manual Dial 10 Whenever agent will do manual call or auto calls it leads will be open in vtiger screen so that agent will come to know to whom they are speaking.
A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign
We are looking to develop a mobile a...integrate specific API from our middle ware platform. Initially the app will only need to be for android devices. The mobile will need to integrate with a telephony platform for sip communication and a messenger platform. The majority of the mobile devices will not have cellular service and will be strictly wifi.
...preferred. Minimum 4.0 mbps download speed and minimum 401 kbps upload ( Check speed through link) [login to view URL] High speed internet, fiberoptic, cable or ethernet. ( No Dial Up or Wireless.) Must have one of following operating systems - Any Windows Basic telephone line with no extras ( i.e. fax, voicemail) VOIP/USB headset with mic for training
I have completed mechanical engineering and passionate about robotics where in have completed Mobile robot using DTMF technology and autonomous robot using Arduino Board from Skyfi labs . Also attending workshop on IC Engine by AerotriX.
hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system
hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly
Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones
...person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not
WE WANT TO DEVELOP THIS WEBSITE [login to view URL]
I need a very simple customisation of a Android based free / open source SIP client(sCIPsimple,Linphone or anything). I want the default values of my server to be populated in place of blanks in configuration forms. So, all you need to do is replace the values and compile and give me.
I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]
Hello. We are looking VOIP Tunnel software solutions to bypass voip calls in blockage area . It is encrypts/decrypt SIP and RTP packets and send voip traffic between soft switches without blockage . We are looking experienced guy who already developed or worked on this .
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
I need some help with finding some leads. leads will be paid $4/lead and payment on Bi-weekly basis. i need to of 6 to 10 agents who can generate me 500 leads every week. Data dialer and Voip will be provided.
...initially provide reporting capabilities on Twilio's SIP trunking product. Using Twilio's REST API, you'll build the following: 1. User auth 2. Dashboard with usage data (much like what twilio provides now, but for the subaccount) 3. Historical reporting - whatever twilio provides data-wise on the SIP trunk side of the house (call date/time, ANI, duration
My project has a mobile controlled lock. Dtmf based Arduino controlled lock By dialling a phone fixed on lock and type 4 setted pin for unlocking
I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a
Android App should generate daily schedule of calls according to preset rules and numbers from list , dial and hangup at time. App shoud autorun with previous settings after reboot. App working principle: * - all X and Y values which will be used below set from user interface Main page Choose program: On/Off Mute: On/Off Show today call duration
...Marketing - [login to view URL] Director(MD) - ONLY WHEN ONE OF THE ABOVE IS NOT AVALIABLE I need to be able to call them directly where possible. Details required are: 1- Direct dial phone number - I don't want the general number where possible 2-LinkedIn Page 3-Personal email address. I'd recommend using a Sales tool like [login to view URL] Some guidelines:
I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any
Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !
Basically converting BICC to SIP and SIP to BICC using OpenSS7 or any other proved open source library. Performance and stability is an important factor
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards
Create and train italian acoustic model based on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should...on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be configured for telephone (8 khz), in order to integrate it in Asterisk with mrcp server.
I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step