Asterisk pbx vtiger työt
Hi, Need someone to configure asterix & free pbx on my server in order for me to make voip calls... few hours job for the right person....please bid only if you did this before...
I need someone to configure a Yeastar VoIP TA410 gateway for use with VitalPBX 4 trunks connected from FXO ports to PBX for incoming and outgoing calls. Will pay max $50. Prefer UAE based freelancers.
i need to integrate whatsapp with vtiger CRM to use it in workflow than sending emails
I need to build a "telephone system center based on Asterisk". (Required in version 17)
Server has a working Asterisk, apache and mysql Would like someone to install A2billing. Connected to Asterisk 11.14.1 currently running only bid if you are experienced I have no interest in FPBX or other third party software, this must be pure asterisk with A2b
We are a UK (our own consultants/developers are based in Indi a where I have a subsidiary company) in India based open source CRM/ERP company, one of the applications that we do a lot of work with is Vtiger CRM and we have a number of customers looking at telephony integrations with FreePBX/Asterisk specifically. We also have a couple of customers who already have telephony integrations in place ...
CRM implementation expert Complete implementation of CRM based on client needs Experience on any one of the popular CRM software such as: Vtiger CRM Sugar CRM Odoo CRM Yetiforce SuiteCRM Should have done implementation for both Sales and Services modules Ability to understand business user requirements, engage with business users to elaborate them further and create SRS and carry out the impl...
I need you to develop wordpress module which will sync woocommerce contacts, orders and products with vtiger crm.
need help with my asterisk setup which we use FreePBX, and some help on my virtualizor and WHMCS setup
Needs to hire Freelancer Seeking Asterisk/Java/JavaScript/HTML developer(s) to make repairs to a web-hosted VoIP API endpoint. Our Product Overview: We have a custom VoIP solution for an open world game. Players on our servers have an x, y, z coordinate and a room r. Players to also have the option to open a web browser, navigate to a URL, and join their voice room (r) with other players. Those c...
Se requiere el desarrollo de una aplicación web que interactúe con un conmutador IP dentro de la red de datos (Asterisk o Freepbx) para realizar la marcación de forma masiva, por medio de un archivo txt o con los teléfonos a marcar e incluir la opción de reproducir un audio editable. es necesario poder especificar la hora de inicio de marcación y fin, co...
El sistema debe estar en php/bootstrap la idea es poder ingresar usuario sip (voip) en una plataforma web y que automaticamente se creen en asterisk (estoy usando VitalPBX). Debe tener: Creacion de usuario Modificacion de usuarios Eliminacion de usuarios lista de usuarios activos y no activos eso para comenzar luego podemos ir añadiendo mas funcionalidades
This project is about "ISP CRM" With freeradius, Asterisk and Network Monitoring Capability which Support Vendors [Mikrotik, Huawei, Cisco, Juniper and Star OS]. Required Features and Modules # Admin Control Panel (Web + Android ) * User Management * Plan/Tariff Management * Finance Management >Invoice generation and payment processing >Recurring billing en...
i have subscribe for the click2call from issabel pbx and the script working fine and i can receive calls from the website but there is no voice i can receive calls from out of saudia arabia i need to know what is the problem BR
Searching for the valuable software house who has exp on developing iOS and Android Mobile application on SIP protocols with the help of Admin panel and Must have experience on asterisk or elastix. Deep Project details will be provided over the PM.
i have subscribe for the click2call from issabel pbx and the script working fine and i can receive calls from the website but there is no voice i can receive calls from out of saudia arabic i need to know what is the problem BR
I would like to get the SIP trunk configuration for goTo and Nextiva sip provider
Currently in Organisation module when we try to send SMS to one record, but for some reason, we cannot select more than 2 fields in the drop field for Step 1 (select number field to send). Organisation record may have the following 3 fields that are numeric: - Phone - Fax - Mobile number We are trying to send to 2 people more than just 1 number, which in this case we would like to send SMS to bot...
Configuration and support for PBX Siemens HIPATH 4000 - for migration and interconnection with PBX Asterisk
I have installed OpenSIPS 3.1 server with UAC, CP and rtpproxy on Debian 9 server with 2 network interface. (Asterisk server - OpenSIPS - Service Provider) Now I am able to establish calls and also registering to my service provider. I need someone's help to fix the rtp traffic and routing.
Necesito que desarrollen un software para mí. Me gustaría que este software sea desarrollado para Linux utilizando PHP.
I need you to implement vTiger crm. i have installed vtiger on my server. need an implementation enginner.
A list of apartments and their owners on a screen. By pressing a phone button, you will call that person. If the person wants to open the door, he presses *9 or something like that. All on a web page. There are JavaScript sip clients. The phone part is by asterisk installed on the same raspberry pie using google voice. [kirjaudu nähdäksesi URL:n] [kirjaudu nähdäksesi URL:n]
Hi, I need to build a sytem that can detect IVR when you call, record a part of the ivr and detect which messages is, with previus record files into the server. The target is to to avoid IVR balance, IVR of blocked sims and detect which record has been detected. We need the build in Python, asterisk.
...are in need of an e-flyer or e-brochure that is easy to email to potential customers. This marketing material should look modern and sleek, incorporate our logo, describe the program, and list pricing (note: the hourly rate for the aircraft should have an asterisk to indicate the rates are subject to
Onyx Flight is an aircraft charter operation that offers a program called the "Onyx Flight Membership" program. We are in need of an e-flyer or e-brochure that is easy to email to potential customers. This marketing material should look modern and sleek, incorporate our logo, describe the program, and list pricing (note: the hourly rate for the aircraft should have an asterisk to indicat...
Need to Upgrade Freepbx from 13 to current and install: SIP Poe Mullion Video Keypad Intercom 8039 8063 SIP Interface Module (IP Relay controller) for Algo door control only. 8063
Welcome messages to be uploaded to a PBX for a client Requirements are as listed. Format: WAV Channel: Mono Bit rate: 8 kHz Sampling: 16 bit If you can not provide them in this format, I can not hire you. The messages will be as follows, actual business name and actual business hours will be provided once I hear some of your samples. Message 1: "Thank you for calling [business name], some...
Hello all, I have Freepbx installed on a VM with a public IP address. I am testing with 2 softphones which can register with PBX and call each other. But there is no audio at all. I would like your assistance to debug and fix the audio issue. I will happy to discuss my setup with shortlisted candidates.
We have a social media app using open source linphone SDK for android and ios SDK for voice. Need voip developer to enable video in existing system and autoscalable solution. Using asterisk on backend rabit MQ
Estoy buscando un desarrollo de modulo de mesa de servicio en vtiger
setup failover for free pbx the task must be completed in 48 hours
i have vtiger function added by previous freelancer, and it has some issue slowing down whole process. so i had to reinstall vtiger module again and like to add function again. *Admin needs option menu to set MAX number of emails sending per every 10 mins, users can't see or access to it. 1. user que table per each user when creating new user, this table also creates when selecting list and...
I have to connect an 3CX pbx to an T-System SIP-pure Trunk (sip-trunk based with 3-way authentification). I want to use kamailio as sbc - but i have no experience with kamailio sbc-config. I need someone to write an [kirjaudu nähdäksesi URL:n] to archive the following: - register to T-System Trunk (3-way auth) - connect to 3CX (sip-trunk) - rtp engine to convert g711 - t.38 and vice ...
Hello I have a centos8 running kamailio asterisk and a2b, the service has chat with file transfers and of course voice and video calls, I have built it with several developers and I have also android custom built clients and apis, Im looking for someone to start installing the required above explained prefferably on debian 10 but 9 would do its a cooperative work between you and me to copy my curr...
Desenvolver uma extensão que ao ser acionada ligue para uma número e faça a conferência com o canal que acionou a extensão.
I need a vtiger expert. i have a wordpress plugin "wp leads builder for any crm pro". This plugin needs to work with crm vtiger 7. Furthermore there surged an email issue, as i can´t send emails currently. The third taks is that i have an email campaign module where i can´t save a new smtp server currently. This problem probably relates to the second issue above.
We are in need of an updated looking soft phone for our call centers. This developer will be using [kirjaudu nähdäksesi URL:n] as the library. Video is not required for this. But implementing SIP Messaging is. We may also require you to call a REST service to autocomplete company directories and such. We are open to feedback for that part of it.
Need to have someone integrate our pbx into Alexa so we can use Alexa device to makes calls from our pBx. There are a few examples of this already on Alexa skills store; particularly as we are not USA based we would like this functionality. Here examples: [kirjaudu nähdäksesi URL:n] [kirjaudu nähdäksesi URL:n] [kirjaudu nähdäksesi URL:n] [kirjaudu nähdä...
Hello, I have installed an Openvox UC 501 Asterisk Based IPPBX at my home. I have a BSNL FTTH Connection with a Landline Number. The landline is currently working through a POTS port on my Syrotech ONT Modem. I want the Landline (VoIP) to work as an SIP trunk on my IPPBX. I live in Raipur, Chhattisgarh, India. Thanks.
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I need a 3cx expert to help me debug the configuration with a patton smartnode 4171, the switchboard in outgoing calls from error 503
I need a 3cx expert to help me debug the configuration with a patton smartnode 4171, the switchboard in outgoing calls from error 503
Me gustaria configurar un pbx issabel en la plata forma [kirjaudu nähdäksesi URL:n] y configurr openvpn y 3 numeros DID adicional
Dear all, I have vTiger 7.3 installed, now the issue I am facing that whenever I import CSF file for customers or leads , and the names are in Arabic , after the import, the names shows as question marks ,
Asterisk Version - 13.35.0 Vicidial Trunk Version - 2.14b0.5 DAHDI Tools Version - 2.1.0.2 We have set up VICIdial successfully and it's running as per requirement, We are facing a problem in the callback. A scenario which running: The agent gets a call and talks to the customer after finish the call if a callback required he will schedule a call, call triggered when times matched with the ...
I have a FreePBX instance and need assistance cleaning up the configuration, tightening security, and troubleshooting SIP connections. Please do not bid on this project unless you have in-depth FreePBX experience.
Hello, We have a 3CX Pbx and I can't reach the management consol any longer. It used to be this URL: [kirjaudu nähdäksesi URL:n] It stopped working sometime between the 7th and 9th of January, I noticed it today on the 9th. I am not sure if we've been using OVH or Google before, I'm pretty sure it's google cloud since I logged into Google Cloud and I have a proje...
Hi Prabath J., I noticed your profile and would like to discuss an upcoming vTiger project with you. Please share your contact details.
