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 should hear the audio file played. Please ...
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.
I am looking for a phone dialer for android devices (android 4 and above) such that it does not show the phone number being called for privacy reasons. The provided application can have a screen with text field where the number to be called is typed and then hitting the call button, the call is connected. The user should see the call information (like
I need an application that can enable online quiz competition on android platforms. First I will need that application to be developed for both android and website [ur...But please bear in mind both will be worked on separetly as I see them to be different projects. So one project at a time. Skills: Android, Java, Mobile App Development, Symbian
I am using Vtiger 7.0.1 and FreePBX(3cx) as phone dialer. In order to prevent data leaks I need to integrate Vtiger and FreePBX and also modify Vtiger to hide our customer's numbers. The idea is to start a call directly from VtigerCRM with a click on the phone icon(something like that) without being able to see the phone number being called. Actually
...skills * Excellent follow-up skills a MUST * Must have previous B2B, Merchant Cash Advance, business loans and & Credit Card Processing experience! We Don't provide the dialer & Voip. We will provide you script & training Payout $15 Per qualified lead on daily basis. Not hourly basis. PLEASE ONLY APPLY IF YOU CAN DO THE JOB AND INTERESTED
Call functions like mute, conference, hold, transfer, and call rec...integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper SIP Webphone
...android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future
...website comprising most of the items in this diagram attached: 'schematic minus BTS [url removed, login to view]' . The diagram should comprise a "MultiDSLA System; DUT UE; IMS; VPP+ (ReferenceSIP Client); DSLA (Analogue test interface); a cloud". similar to '[url removed, login to view]'. diagram should be in a style similar to the 3 examples given be...
I need you to develop some software for me. I would like this software to be developed for Windows using .NET. Sample C# Project to :1. Connect to SIP trunk2. Make a call : play a file when pickup , hangup after playing , run event when hangup(by system or user) or no answer3. Receive call : play a file , run event when hang-up or no answer (by system
Hi, I'm trying to setup a Debian Jessie SIP server with Kamailio. Everything works but the TLS handshake doesn't complete. It stops at the client handshake, so the server doesn't send it's certificate. I would like someone who has experience setting up Kamailio with TLS and unix server administration. The deliverable would be to let me know
Must be experience in Open Source IP-PBX development product. Setting up the infrastructure, build key features and testing. Works on Debian® 2.8 distribution, some of the key FOSS components that support the unified communication and management functionality are Asterisk®, FreePBX®, Chan_Dongle, and RaspAP Web GUI. Build a IP-PBX product that allows users to make VoIP calls to...
It's simple I will rent a Linux Server (Centos u Ubuntu) and I n...(Centos u Ubuntu) and I need to install a dialler in it. The dialler is an Asterisk based one and all it's suppose to do is send call automatically to a telecom server by using a SIP account with 30 concurrent calls capabilities. The account has already been set in the Telecom Server
...I am new on ASTPP. I use the link([url removed, login to view]) to manual install the ASTPP. I configure the trunks, rates, sip, gateways, etc. All are configured. When I try to make a call, show me error on fs_cli and the call hangup: 2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE: IM002
...create origination rate table and termination rate table 4. create trunk and set up routing so that origination carrier calls cam be terminated to termination end. 5. create a sip user account on new customer and route his calls to trunk of termination carrier. test and make sure all calls connect properly. give me a walk thru of the steps taken to