Asterisk PBX Jobs
Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.
Punësoni Asterisk PBX Developersmini POC WEBRTC-to-SIP architecture backend buildout for SIP server to webrtc (kamailio or janus) docker build setup to Sample this SIP-WEBRTC (to enable websockets) client= webrtc sip register=SIP server does RTP have to flow through kamailio or SIP-WEBRTC as well trying to model off this [identifikohuni për të parë adresën URL]
we want to separate Freeswitch server on our bigblue button sever
I require installation of goautodial 4 in vultr (VPS), with the configurations, Trunk, Users, Recordings, Webrtc for remote agents, everything required for call center
I want to achieve the following: 1) all incoming calls into PBX should show as popups in vTiger. If they are existing contacts, that contact details should be shown. If not, then new lead should be created. Outgoing calls from the pbx should also be logged into the crm. Click-to-call in crm should be functional. Can all this be done? Please revert at the earliest so that we may get this going. tha...
I am running ISSABEL 4.0.0 with asterisk 13 as well as vTiger 7.1 free edition. I want to integrate the two.
Se requiere un profesional con experiencia en AGI php para plataformas Asterisk para crear y/o modificar un IVR transaccional de cartera. También debe tener conocimientos básicos en Mysql para actualizar sistema de carga y reportes del AGI php.
we are looking for a goautodial expert for long term job
cisco phone: 8841 Free PBX: [identifikohuni për të parë adresën URL] Static IP of LAN Our office
i want cloudbase ivr telephony solution where we can redirect to different departments
To make custom Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience required: 1) LibreOffice Calc (Spreadsheets) 2) Software to change to pdf without loosing calculations 3) Asterisk PBX 4) Linux 5) MySQL DB 6) PDF 7) PHP 8) KVM qcow2 (for disk management) To show invoice and Credit Note Templates, we use backgr...
hello, we would like to create a web softphone for our asterisk freepbx server. the page must have username and password to access the web phone. on the same page there must be information such as, campaigns enabled, calls waiting, calls answered, and average conversation time. all on one page. then we would like to have a control panel where to enable users to the various campaigns / queues
looking for developer in WEBRTC expert. Allowing one device(the "master") to broadcast live audio to multiple "peer" devices over local WiFi. As the app should work over WiFi LAN only (with no internet access), the signalling sever "SIP" SERVER. Thanks.
looking for developer in WEBRTC expert. Allowing one device(the "master") to broadcast live audio to multiple "peer" devices over local WiFi. As the app should work over WiFi LAN only (with no internet access), the signalling sever "SIP" SERVER. Thanks.
we would like to have both the sip client and the telephone bar where the user sees the code where it is inserted and where there is the possibility to pause the sip
We need to have a auto dialing system built in php which can work on voip and with internal chat window for team just like go autodial vc dial
we are planning to setup our branded IPBX based on asterisk or freepbx
I would like to add / update existing extensions in my FreePBX.
trying to be able to use SIP native for android . but also the same UI/layout interface if on the PWA it will work the same but webRTC not sure what libraries to code into. device Mac based ID sends to API built with ionic framework/ angular. not sure exacty what to use [identifikohuni për të parë adresën URL] jsSip/ kamailio or what not.
We need an experienced Asterisk, PBX, within Odoo to help set up the Asterisk PBX. We need someone with proven track record who has done this before. We also need your availability and comfortable with screen sharing and working together on a live platform to install and you provide support. IF you are not willing to support or work with us Live, please do not apply to this job.
To make templates for Invoices and Credit Notes, to be filled out on the Internet. Must be changed to pdf without loosing calculations before being uploaded on disk 3 of our server (KVM for disk management). See attachment
Required expert in Cisco phones to setup custom features for the phones. We are using Elastix pbx server.
We are a telecom company. We deal also in VoIP and in SMS services. We are looking for someone that has some experience.
Integrate voice api, or build from scratch, Idea of application is like a dating app, only have voice chat for the moment, With paid plans in app Random strangers call with online users, Choose genre hit search and talk Simple application idea Could use ready api from any provider such as Twilio etc... Please come with ideas Thank you
Hi Friend, We have stuck at our PBX to receive more call, the problem is description here: [identifikohuni për të parë adresën URL] My Server is : 12 VPU, 14G RAM, almost free when error occur We try so many way to improve the PBX but not suceess I need an export support me config to receive more call from PBX The info of system: - FusionPBX - FreeSWITCH version: 1.10.2-rel...
Necesitamos integrar un webphone a nuestro CRM el cual esta programado en PHP7, los requerimientos serian los siguientes. 1-Posibilidad de loguearse en conjunto con el usuario del CRM 2-Posibilidad de conectarse a central telefonica por el protocolo de comunicaciones sip 3-Posibilidad de realizar Click to call 4-Integración al CRM para realizar y recibir llamadas
1. Android:Bugs to be fixed and updates to add: a. Contact- app crashes on some people's phones usually while loading contacts...some exceptions were probably not caught properly. Also contact should have search icon to make contacts searchable b. On changing the layout from portrait to landscape, it crashes the app c. App crashes when users click on the chat and try to chat sometimes. W...
We are currently looking for an experienced VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, and Linux experience and familiar with coding capable of setting up an Interpreting/call center solutions platform with the following requirements: • 5,000+ Interpreters • 300+ Ring Groups • IVR • Phone Calls • Video Calls • Text Messages • Accessible from Smartph...
I have account in [identifikohuni për të parë adresën URL] and i have ip phone yeahlink t-48g i want to make calls from the ip phone throw thee mobile phone line (note 8) so the caller id should be my mobile number. the basic for all in & out call is my mobile line. the second thing how to i setup when i want to get all call only in ip phone. you will get access to adm...
I am a small VOIP business , several of my customers have been complaining that their DID's cannot be reached by callers with specific carriers... its a very strange issue and I have no way of troubleshooting it as there are no logs for this anywhere. *** I AM ONLY LOOKING FOR SOMEONE WHO HAS DEALT WITH THIS ISSUE BEFORE WITH CARRIERS IN THE US***
Kamailio Server setup on ubuntu or Centos connected with linphone sip client .use UDP Protocol
Eu preciso de um novo site. Eu já tenho o projeto, eu só preciso de você para uma página para o meu pequeno negócio.
