Questions tagged [voip]

Voice over IP (voice over Internet Protocol, VoIP) is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

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Secure FOSS alternative to Skype on Linux & OpenBSD?

Criteria: Makes audio/video calls Encrypts the whole traffic (using good encryption) Is cross-platform (including Windows 7, etc.) Runs on modern Linux distributions (Fedora, Ubuntu, etc.) Runs on OpenBSD Does anybody know a good Free and…
LanceBaynes
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How can I set yate to be my default tel: protocol handler?

On this site I clicked on the support link and a phone number popped up. It was formatted via the tel: protocol, and as such was underlined and highlighted like a web link. On my phone, clicking on such a link opens my default dialer and places…
virtualxtc
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Jingle CLI client / Jitsi replacement / VOIP

I am looking for a Jingle client for the command line. Currently, I use Jitsi, but my dreams of not needing the GUI for more than just web-browsing are not realized due to my reliance on this program. Does anybody know of a CLI client that…
Aviator45003
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Easy voice talk between 2 computers in 2 countries?

My plan is to enable 2 sets of unsophisticated users (children and their grandparents) in 2 different countries to talk easily. In practice I see it as: A quiet Linux computer is on all the time in both places. The user at one computer should be…
r0berts
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How to setup QoS on router (prioritize VOIP and web browsing over downloading)?

I've got Asus WL500GP-v2 router with renewed Oleg's firmware installed (http://code.google.com/p/wl500g/ , the stable version). I would like to setup some Quality of Service to ensure all users are able to browse web or use VOIP when one of them is…
Joudicek Jouda
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how do I make a SIP phone ring with sipsak?

How can I use sipsak to simulate a call so that there's an actual ring, or possibly even a sound is played? thufir@mordor:~$ thufir@mordor:~$ sudo sipsak -v --usrloc-mode --from sip:demo_alice@localhost --password 123 --sip-uri…
Thufir
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Audio file as an audio input device

I want to play an audio file over VoIP in order that people can listen to it with a phone call. One way to do this would be to set up the audio file as an audio device where the audio starts playing at a specified time. There are better ways of…
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VoIP dialer for SIP

I have some Java, so am biased in that direction, but not overly so. None of the Java…
Thufir
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Asterisk, GSM-FXO gateway example of usage

Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone stations?
ThatsMe
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(IP Telephony, VoIP and Video Conferencing) Ekiga configuration for LAN with same subnet

I am looking for an Open Source Voice Chat Application, like Skype, but within an isolated intranet (no Internet is available). Is there any application which can run on Scientific Linux or CentOS? Client-Server will be OK, but our need is Client…
TPS
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Asterisk: how to change the CLI color prompt?

The Asterisk wiki says that the color of the CLI prompt can be changed with %Cn[;n]. ie: export ASTERISK_PROMPT="%C31[%H]: " but when I use the export above, I get broken prompt [1;31m[voip]: [1;0m, where voip is my hostname. My terminal is color…
400 the Cat
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How do I fix this error: "ipset v6.11: Hash is full, cannot add more elements"

When I run the update script for voipbl manually, I get this error from ipset: ipset v6.11: Hash is full, cannot add more elements. I am running it manually because some IPs that are on the blacklist seem to still be getting through the firewall.…
d10nte
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How do I prevent Viber from turning my microphone volume too high?

I am using Linux Mint 17, and I installed Viber to use as an alternative to Skype. I got the Viber client that is available on their website: https://www.viber.com/products/linux/ Whenever I start a Viber call, my microphone volume automatically…
Dom
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How to make Skype conference calls and screen sharing in Linux

Now that Skype got discontinued for Linux and only the 5 beta version is avalable, Linux users have a problem, which is how to communicate with Mac and Windows users. There have been lately a lot of questions regarding the installation of Skype and…
TomCho
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Asterisk and NAT: SIP and IAX registration failed on remote connection behind NAT

I set up two asterisk servers (on Fedora) in different networks. My goal is to make a call from softphone (on windows lite with ip: 192.168.20.3) to the asterisk server 2 which is in the other network (ip:192.168.10.2). But the problem is in…
Y. Dabbous
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