Asterisk chan dongle trabalhos
Need to setup an Asterisk, FreePBX based VOIP system with IVR and with some other special requirements. PLEASE! Bid only if you have awsome experience with Asterisk, FreePBX, and automated phone surveys. You will need to install, configure the whole system over SSH connection, the system will use multiple USB modems as GSM trunks. IMPORTANT! I don't want to waste your time, please do the same thing. Our budget limited, we have some system admin experience, we know what we ordering now, in this case, please bid with an affordable price!!!! Thank you.
After creating freepbx extensions from mysql and running fwconsole reload, I noticed I am unable to receive incoming calls, except I open the extension again on freepbx UI, click on submit and applyconfig which is an overkill. I need to replicate this action from the freepbx terminal(probably running the php script on individual extensions. Any other way is also fine
i work on chan_mobile with asterisk by bluetooth i am looking to work on cable instead of bluetooth just i need the way how to do that
We need to be able to 1. allow the user to call the lead phone number, call the contact in opportunities, call the account 2. call a campaign All this should be used via vicidial with a combination of a webphone/soft phone. Since vicidial is based on asterisk we could use direct connection to asterisk also, but logs etc must be available of course.
necesito soporte de telefonia y tecnologias de colaboracion basado en telefonia VOIP, asterisk, freepbx, issabel
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.
Buenos días, Buscamos programador Linux / asterisk. Sería para proyectos ya en curso, en este caso pago por horas. Y para proyectos completos precio negociable. interrelación de diferentes equipos y tecnologías
Buscamos programador por horas e incluso poryectos Linux / asterisk. Urge Se pagará según valía de candidato
...confirma el cliente que desea contratar el servicio. * Desde la pagina de administracion poder pasar con un click al panel del cliente ( sin autenticacion ) * ventana emergente de las llamadas entrantes con datos del contacto, si existe, a la extension. Ya existen soluciones comerciales de este punto, pero necesitamos adaptación a nuestro proyecto. Se trata de integracion con Asterisk, aunque usted no necesita conocer Asterisk o el protocolo SIP, dado que el programa recibirá los eventos de las llamadas entrantes con los parametros necesarios, como tenant y numero llamante ( caller id ). Usted puede elegir el lenguaje orientado a eventos que conozca. Si lo desea, aportamos el servidor para el desarrollo. Sabemos que está desarrollado y se come...
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.
Нужен образ docker с настроенным Asterisk который проигрывает заданный audio файл в зависимости от dtmf кода
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.
NECESITO UNA APP QUE FUNCIONE CON EL SISTEMA MBILLING softphone asterisk
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...
