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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.
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Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gateway converts the sip/iax...
Hi, we want to find a software that can use in Linux system that can get caller_ID, status ,log from analog inbound telephone. This software should be able to sync the status of the phone (on call, processing, stop, miss) to other services in the same local system. Now, we are running our local Tomcat server by raspberry pie.
I need a BYOD platform for VoIP where we can make it easier, cheaper and simpler for Small businesses to connect any VoIP-enabled device and be up and running in few minutes. It must be a Zero touch platform. Clients must be able to easily signup and allocate their own calling plans, load airtime, create extensions and allocate, etc. Here are the most basic features this platform must cater for: Uncapped/Capped Calls / Easy Usage IOS & Android App Integration / Call Groups / Superb Voice Quality / Voicemail & Call Routing / Cloud Hosting / Basic or Advanced IVR / Live Chat / International Extensions / Quick & Easy Set-up / Complete Scalability & Flexibility / Video Conferencing
Hi, need to develop an application that analyzes presence of tone in early media and makes the decision to drop the channel. It must support multi channel dialing (eg: when dial application is called and it has multiple dialing destinations)
Review long distance carrier toll free routing tables, switch records and call detail reports to determine reason for 75% drop in toll free call volume on one 800 telephone number, This is not SIP or VOIP. This is traditional long distance service and requires expertise in telecom switching for long distance carriers.
Hi, We need assistance to set up Elastix last version 5 1. 7 internal extensions 2. 4 fxo ports with analog lines (sangoma a400 card) 3. Dedicated outgoing calls from extension through the correct fxo port 4. Incoming call to dedicated extension on chosen fxo. 5. Fax to email Reach me by whatsappp 198o322872two
We are currently building a Kamailio SBC which will be as Sip Proxy for our B2B, but we are facing a bug. Seems there to be a misconfiguration in Kamailio because when we are using a dispatch request, re-invite and ACK is not working to be sent in right way, because the re-Invite is sent back to dispatcher list which is making it lost and it is not forwarding to callee. This issue is mostly present at ACK and BYE. Hope some one can solve this and fix this SBC.
Witam, Interesuje nas pomoc przy ustawieniu centrali freepbx (z gui). Potrafimy zainstalować, wstępnie skonfigurować, uruchomić do działania. Nie potrafimy jej doustawić, tak aby połączenia wychodziły z odpowiednich zakupionych numerów, itp. Chcemy to zrobić z pomocą ekspertów własnymi rękoma (aby się nauczyć). Przydałoby się na to zapewne kilka godzin ze specjalistą, który nam wskaże co robimy źle.