Chan dongle asterisk ussd jobs

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

I. Graphic interface to create IVR flows with the capability to integrate external systems such as data base, web services and APIs. 2. Data Base: MySQL 3. Report for all interactions by phone number, hour, date.

Looking for someone with skills in: ReactJS Redux PostGresSQL (Sequelize) Pro in Asterisk and WebRTC Excellent Web responsive developer COMMUNICATION IS IMPORTANT TO ME, GOOD, RELIABLE COMMUNICATION - !IMPORTANT WEBRTC WITH SIP via SIPJS, JANUS, or SIPML5 - !IMPORTANT SKILLED AT MAKING DIALPLANS IN ASTERISK AMI AND ASTERISK AGI - !IMPORTANT NEED SOMEONE WITH VERY GOOD ENGLISH - !IMPORTANT The application is in the process of being built. I need someone to take care of the additional tasks. I need someone who can dedicate their time and provide clean and organized code. 1. "Build in call routing into the groups. Inside the SMS Group settings, when a call comes in we need to be able to decide if the call will: - Be forwarded to ring the last agent that spoke to ...

CRM company looking to add a multi-tenant PBX solution. Currently using Asterisk but looking for something better to handle many simultaneous customers and to API with current software (php, mysql, ajax). Require the ability to add, edit and remove SIP users along with codecs, voicemail and call history. Will use Twilio for origination and multiple wholesale providers for outbound. The engineer awarded must create the AWS linux instance, configure PBX and assist in providing API information. Additional maintenance work will be needed.

Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

We want a Based-Asterisk system as: -5 routing devisions: Premium, Gold, Standard, Dialer, Special, -I have to upload as max as possible of rates, from carriers, for each devision, and not set the rate for client, but to set a percentage of revenue from the carrier rates, like if in Premium, i choose to get 20%, it should find the best carrier for each destination and give the price to client + 20% -Special Devision is special for each client, with different destinations and offers for each client to be used upon a special deal only for him, -Auto blocking or auto price increase with a set of % for bad ASR/ACD to be choosed, -Online payment, using PayPal and Stripe: sipmle just API set and minimum/maximum per payment. On payment page, show also bank details, with the account number ...

Make software with A2Billing, or similar open source software, using custom made invoice Templates, to be filled out on our server. The Templates are made with LibreOffice. Server and PBX (Asterisk) are on different machines and networks, with different IP addresses ! All is Linux - no MS ! You must be very familiar with Billing software and Asterisk,

we need to have a predictive calling to grow the volumes on our call group what is the cheapest and best solution we can use? (OK, I know asterisk and did this some 20 years ago, but then you also need a VOIP provider) does someone have a cheap solution here we could install? (we are a startup so we do not have much money)

Install and configure and integrate with GSM gateway and FXO Gateway on cloud and On prmsis and bridge them. Create Ext,IVR, Quw similar to existing 3cx . STABLE SNG7-PBX-64bit-2002 Release Date: February 2020 FreePBX 15 • Linux 7.6 • Asterisk 13 or 16 Supports UEFI and Legacy BIOS booting

On a system running Asterisk that has iax trunks you can run asterisk -rx "iax2 show peers" at a command line and it will produce give the output found in iax2-nodes file attached to this project. I need a php file that will scrape the data out put by Astrisk and produce a file rpt_exnodes. You can find an example of the output i am looking for at This file will be used for a non profit ham club not a commercial deployment.

Looking for a developer familiar with Asterisk Opensource PBX software, to assist in a build out of a multi-tenant PBX on that platform. Please apply with your experience with Asterisk, and availability.

Make video webcam streaming website with credits: simular like this website: Integration with: Asterisk - WebRTC - Mysql - WP - PHP

We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server

VoIP developer requires good knowledge of voip development free switch and asterisk developer .

Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distribution C. Centos 5.8 or 6 7. Server A to Server B voice traffic wi...

Asterisk 17.4 is downloaded and extracted, but needs to be properly installed onto a server for us. PJSip config to a SIP provider, the ability to provider music/advertising on-hold, a couple simple queues. We do not want freepbx, please do not suggest it. Just install asterisk and, as needed, help get config files setup.

Both Chrome extension and windows application will provide connection to Asterisk AMI for click2call dial and popup based the caller callID. The chrome extension will change all page phone numbers (based on a phone pattern set in option) to click to call link with a phone display. The extension should provide same capabilities as FOP2 The windows application will provide DLL for other application to get and set events.

1. There are multiple asterisk servers, each provide services to multiple tenants. 2. From a main centralized server (panel server), I need the option to load file (word, pdf, tiff, txt) and send it to the relevant asterisk server. 3. Current asterisk servers has a local webpage to load and send fax. 4. panel server has full access to local asterisk servers and open AMI connection.

i did install the freebpx but i have nothing but problems, i need someone to help me with it, secure it and make sure its working to its optimal condition. .

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

Install Asterisk in the same 3cx Server make a General SIP trunk with DID number from one of your internal extensions, then add to this trunk DIDs for all of your internal extensions; You need to add inbound rule for each of the trunk DID and configure it to route calls to corresponding extension number. Integrate on Vtiger and zoho CRM , noting that Vtiger is on cloud server and thus the Vtiger and Asterisk Server are different location, then MySql need to be open to connect Vtiger and Asterisk.

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

Hello, I looking for someone to help us to build Asterisk IVR/PBX with call blast dialer and predictive dialer.

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

I registered for an online sip provider. When I use their app, I can add caller-id's by verifying phone numbers by sms. Without verification, changing to any custom phone number, is not possible. When I use the same SIP from my own asterisk server. I am able to change the parameter CALLERID(num) to ANY value, and without verification, the caller ID shows up on calls. I would like to use the provider directly, without integrating it into my own asterisk server. I have an open source iOS app = linphone. I can set custom Headers etc.., but I have not found a way to set a custom unverified number as the caller ID. Objective: Provide information which will allow me to call from any caller ID using the open source linphone swift iOS app.

I need integration of opensip over asterisk. NOTE :- DONOT BID UNTIL YOU DONOT KNOW ABOUT OPENSIPS

We are looking for someone who have a sound experience of Asterisk and Json. He or She must possessed some sound quality which should adhere to our standard

hello I have asterisk ip pbx and i need someone to setup call us web rtc button on our website please see the image so you can understand our requirements

We need someone with a lot of knowledge of asterisk and webrtc. We are looking for a partner who can handle the ongoing development and error correction. We are only looking for people where we can see great reference on the asterisk

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all det...pay clients for their incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" includin...

Integrate my FreePBX system with Zoho CRM - Phone Bridge See more: zoho c, zoho, bridge, asterisk pbx system, pbx phone system, crm pbx, teamspeak phone bridge, crm zoho user guide, crm zoho magento, asterisk phone system, integrate crm asterisk, adobe connect phone bridge, php crm zoho, freepbx phone, asterisk pbx crm, sugar crm zoho crm, integrate asterisk pbx, phone freepbx, asterisk bridge, a2billing freepbx integrate

Wish to have a integration of a Asterisk PBX like FreePBX or something with ERP Next. We should have incoming call popup with data being picked up from the ERP Next CRM Module for a existing number and We should have a option to save a un-known new number as a new Lead or Save to a existing customer / supplier / other contact similarly we should be able to do a click to dial from within ERP Next

I want to learn opensip integration in asterisk . I already install asterisk server . Thanks in Advance

I want to install opensip in asterisk... Note :- Donot waste your bid until you donot know about opensips

This project is to write instructions to make special Templates with Scribus, to be used by total newbies. The Templates use a background image as base, with sometimes a spreadsheet for Cell addresses, used by Asterisk PBX.. Knowledge required: Scribus, LibreOffice, Asterisk, Java Script, etc. See attached Start Page. Details to be discussed. NOT FOR BEGINNERS !

...my signal. I broadcast in TVWS. Which is UHF. 470 to 690 mhz. I do not desire an antenna that will encompass all channels in that range as I know that creates bulk and size and little gain. I am willing to take designs for antenna specific for the channels (frequency) that I am using. The frequencieschannels that I am using are as follows: 473mhz (chan 14.) 641mhz (chan. 42) 671mhz (chan. 47) 689mhz (chan. 50) 503mhz (chan. 19) 581mhz (chan. 32) I broadcast in the Vertical and Horizontal plane. on my tower. Again. Quick run down. My tower. Broadcasts at 30meters.18 dbm. 14dbi antenna My client radios range from 6ft to 20 ft (5-6 meters) with 1-5 dbi log periodic antenna) 18dbm tx power. Each client radio has two antenna. My client sites...

Message is D9775D0E0A8FC7EFBA9B0E1287D961F7B80C4ACF41D52911E682C15C2069798C0ECBCF65D0DBCD4EBBCBA0B71C549F97416190BC3C4687E5E732681C969341617B38CD0E8BD9659059FE6E83C2EE3C88991F87DB6F719A5D06CDE9EFF13A3DA7BB1A comes in via ussd I need for someone to decode it and tell me what format it is in so I can decode myself Do not bid unless you know the answer

I need to configure Kamailo SBC on Ubuntu 18.04 to connect multiple Microsoft Teams account in the same SBC server. Require complete / usab...04 to connect multiple Microsoft Teams account in the same SBC server. Require complete / usable and repeatable documentation for: - Install - Configuration of Kamailio - TCP / UDP / WebRTC Proxy passing through to Asterisk on Private IP - Registration pass through to Asterisk with rewrite to present sourceip as reg contact on ast. - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk MS Teams 3 <--> Kamailio SBC <--> Asterisk ...

Hi Alexander, I noticed your profile (VOIP Asterisk freepbx) and would like to offer you my project. We have a conference call going via an Asterisk server with (let's say) 100 users. We need to record individual users in 15 sec intervals -> send wav files to Google for speech recognition -> process the returned text by searching for a few bad words only -> if at least one bad word is found then write to an internal db and trigger a mute function on that user. Is that something you would be interested in? If yes - how much would you charge and how long would this take? Thank you, Michael

Need software to control a USB device to turn it off and on. It will be for a dongle that is connected to a USB connection, we need to be able to turn it off and on in order to change the IP address. I am sure this will be a simple project for someone.

I have dump file from type of dongle wibu box you can create dongle emulator for activate the software

I have dump file from type of dongle wibu box , you can create dongle emulator for activate the software

I have dump file from type of dongle wibu box, you can create dongle emulator for activate the software

We need to downgrade our Huawei Software version. From Software version: Web UI version: To Software version: 22.328.62.00.1080 Web UI version: 17.100.19.04.1080 This will be a simple project for someone that knows what they are doing. The issue with the new software version is that we cannot change the IP address unless we unplug the dongle we are using a remote desktop set up so it is impossible for us to do this.

Need an Outlook 2016 plugin which allows for CTI in an asterisk based call center

i have need a dongle emulator for a software whose i have broken the dongle this work is perfectly legal i can provide photos of dongle and software in the box. i own a dump of the dongle

I have Asterisk dialer for outbound shortcalls compagins its dialing perfect all calls issue what we need to fix which see we are getting if we are sending 10calls per second it will dial for a min then pause for 50sec then restart dialing i want keep sending calls at 10cps without take pause please bid only if you understand issue we both can save time of each other