Cloud Based VOIP Auto Dialer for Voice Broadcasting with 500 Concurrent channels no need of agents required, only ring and play message , should use VOIP SIP routes , multi campaign support [url removed, login to view] ID [url removed, login to view] upload [url removed, login to view] time setting [url removed, login to view] upload [url removed, login to view] campaign
I have a ByteSaver Installed Over Linux Server need to change The ByteSaver Sip Ip Inside To Another Sip IP, Need Someone Familiar With ByteSaver VOIP,
...for iOS and Android. The call between SIP server and the webRTC client will be within VPN and the client has to then invoke a web based application using device internet passing on the caller number and some other parameters: The mobile application will facilitate both inbound and outbound calling on SF The inbound call lands on Sip Server Routed
SIP Program (Really simple) Create, edit, delete users & each user has one-more parameter named URL. when call inbound to user the program knows send GET request with callerid to specific assigned user's URL. GUI is not required, you can make it with NodeJS or C# or whatever. it should be SIP server for anything. (the program)
...need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The devolpment should run under GNU/linux (Asterisk,etc). The implementation should return the correct call error codes to the SIP backend, i.e. CALL
A mobile dialer app for modern mobile interfaces that should work with any sip voip server using sip protocol and support all the standard codecs, consuming low bandwidth, voip dialing using wifi, 3G/4G and have the functionality to use local minutes (as calling card). user friendly and quality voip calls. User may registered through his verified mobile
...webpage, and the colour of most text. List will be provided as a spreadsheet (Excel or Google Docs), to be filled in with Hex Colour codes, captured using an application like Sip. We will provide several examples where we have captured the colours already, and will check in and verify work after the first 5-10 to make sure that you are getting the
I need a sip-phone (IOS, Android, Windows) able to register into Asterisk and peer a SIM in a GSM gateway. Sip Client should be able to send/receive voice calls, SMSs and USSDs. Sip Client should be able to top-up the peered SIM.
We have conversations (see githup) as basic and inside this we need some modifications. We run our own xmpp server and separates sip server with csipsimple. By api the registration in conversation is there one sip and xmpp users. Under the conversations user can call other user in his contact list. For this there is already some working examples.
I will give access to a linux box, you can install asterisk along with any other services, I want a web interface, where i will copy paste 200 phone numbers, also there should be option to upload a voice record (audio) , once i click play button. Each of the 200 numbers should get a call from their own numbers (spoof) simultaneously and all of them
Skill Pre-Requisites: Knowledge of WebRTC Gateway and not WebRTC web-client. Have developed SIP Servers applications (SBC or PBX) Good knowledge in WebRTC, Web Sockets, HTTP Knowledge of Asterisk server and Jitsi Media. Experience in product development and system engineering for WebRTC. Knowledge on JAVA will be good to have (Not mandatory)
We have our own sip / xmpp server and we have compiled the latest build of csipsimple and Linphone. We want a secured voice over zrtp. This is a peer to peer encryption but i dont get it work, not under csipsimple and not under Linphone. I need support to get this work. Linphone and csipsimple can be downloaded on playstore.