Voice Over Internet Protocol is a name of a family of technologies that allows transmission of voice over an established network connection. What is usually discussed, though, is Internet Telephony - bidirectional voice (and lately video) transmission over public networks.

learn more… | top users | synonyms

3
votes
1answer
622 views

A good softphone for OS X

I'm sorting out a VOIP setup for the office that I work in (there's about 50 people, mostly macs), and I'm looking for a good SIP softphone that would work with an Asterisk based PBX, and I figured ...
2
votes
1answer
38 views

Bluetooth headset does not switch to VoIP

I ran into a problem when using my Bluetooth headphones on Windows 7. They support both the A2DP and the Hands-free profile, however it seems that only one of them can be active at a time. On my ...
2
votes
1answer
6k views

“Caller rejected because extension not found” in asterisk

I'm an asterisk newbie and I've just installed it on my ubuntu 12.04 server with sudo apt-get install asterisk. I'm following the tutorial for asterisk from here: ...
2
votes
1answer
224 views

VOIP over double VPN isolation system = bad call quality, solutions?

First of all, I'm not sure that this is the correct Stack Exchange site to ask this question on. I originally asked this on Information Security but was told that because this question involves a lot ...
2
votes
1answer
1k views

Garbled VoIP over VPN, mainly on conference bridges

My office has a large Cisco UCM setup, and I work primarily remotely with a physical Cisco 7940 phone. I've battled some quality issues before, which turned out to be a bad switch port. But mostly ...
1
vote
1answer
70 views

Sending keyboard presses to minimized rdp windows

Is there a way to send key presses to a minimized rdp window? For example: If I was rdping to another computer that had a voip program that uses a push-to-talk key (for example, mumble) is there any ...
1
vote
1answer
125 views

Using Ekiga from a hotel

Since Skype crashes whenever someone signs in, signs out, or sends a chat message, it has become unusable and I need to change. A post at AskUbuntu lists Ekiga as an alternative. When I try to use ...
3
votes
0answers
1k views

getting pbxes, sip2sip, ipkall all working

I was able to make a call from my landline to my ipkall number and have it ring on my android device running sipdroid. Just once. Since then, when I call my ipkall number, I get "The person at ...
2
votes
0answers
295 views

Good quality audio conferencing without a headset

I work at home and am doing a lot more video conferencing than I used to. I also use a softphone on the computer for most of my phone calls as well. Particularly when video conferencing I would like ...
2
votes
0answers
32 views

Videoconferencing software that degrades framerate rather then resolution

Is there a video-conferencing software that can be set to degrade framerate rather then resolution when the intervening connection is slow? I do a significant amount of remote collaborative work. ...
1
vote
0answers
22 views

Ekiga softphone shows as “registered” — but is it actually?

Is Ekiga actually registered? How would I know? I think Ekiga connects, or at least it says "registered": But, when I exit Ekiga, and use SFLphone, it fails to register: While sipsak shows a ...
1
vote
0answers
37 views

Error: “Channel not in Stasis application” when adding channel to a bridge

I'm working with Asterisk ARI. When I create channels via ARI, I specify application name and I can add those channels to a bridge with no problem. But when I use other client (like Zoiper) to make ...
1
vote
0answers
230 views

Ubuntu + Asterisk = Registration from '“100” failed for '10.0.83.202:51966' - Wrong password

I have set up a simple Ubuntu 14.04 just installing Asterisk (apt-get update && apt-get upgrade && apt-get install asterisk). When I try to log in from my client (Telephone for OS X) ...
1
vote
0answers
118 views

FreeSWITCH simple configuration for audio call between two computers

I have two computers and I want to make audio calls between them. Each one of them is behind a different NAT. I am an absolute beginner and I have no idea what I need to configure. After reading and ...
1
vote
0answers
172 views

How to break Lync / Cisco IP Softphone interaction when placing a call from Lync?

I have the following VoIP configuration on my Windows 8.1: a Lync 2013 client (connected to an OCS server) Cisco IP Phone (for PSTN calls) a speakerphone (speaker & microphone, namely a Jabra ...
1
vote
0answers
137 views

Stream windows audio over network and send back mic sound

I am a little lost as to how I go about implementing this kind of setup... We use SIP Softphones in our office, I would like to be able to, from another computer on the network, listen into the call ...
1
vote
0answers
679 views

Best way to block users from connecting to VOIP (Teamspeak)

In a teamspeak that I work with, we've banned users from comming back and they just change their IP and come back on. We've been searching for a way to keep them out even if they change their IP. Do ...
1
vote
0answers
196 views

pfSense allowing inbound sip phones, but not sip trunk

I am currently trying to get an inbound trunk working on a remote Asterisk machine. The ip phones are remote and connecting just fine through port 5060 (I can make internal calls to voicemail, other ...
1
vote
0answers
51 views

How can I blacklist a program from accessing a mic or volume controls?

In this particular case, I have a program that ignores settings for mic usage (eg, it always-transmits even when its own push-to-talk and such is set, and causes all other program's volume to be ...
1
vote
0answers
670 views

how to reduce h264 annex B NAL unit sizes to avoid IP fragmentation?

I'm trying to stream a video over to a terminal using RTP. The video is coded in h.246. I'm using ffmpeg to extract the byte stream according to h.264 annex B : ffmpeg -i test.h264 -vcodec copy -vbsf ...
1
vote
0answers
224 views

How to run secure voip traffic without using vpn

I use quintum tenor for voip traffic using VPN SSL Tunnel. VPN server is based in another country. VPN used enough bandwidth, and we need to reduce bandwidth consumption. Is there any SECURE way to ...
1
vote
0answers
162 views

Start call on Cisco Voip phone with an http link

I don't have any previous experience with IP phones. I'd like to know if there is a way to make a call with a Cisco IP Phone (model 7912) clicking just a web link. Actually the phone is connected to ...
1
vote
0answers
170 views

using zfone in practice - how do i force gtalk to use SIP?

Recently i got a need for a fully encrypted and safe communications. Believe it or not - there is no simple solution. Criteria for needed software include: 1. Encrypted 1 to 1 chats (text). 2. ...
1
vote
0answers
743 views

Using Asterisk with multiple phone lines (PSTN)

I've had experience working with Asterisk over VoIP. Did pretty complex stuff with it as long as it had ethernet cables and internet connected. I've gotten used to connecting the box to an ITSP and ...
1
vote
0answers
365 views

Voice activated and silence deactivated audio recording software

I plugged my PBX into the microphone hole in my PC's sound card, in order to record all activity on my PBX in MP3 format. I am using the software called "Recordpad", by NCH. The problem is the ...
0
votes
0answers
3 views

Interactively play sound clips in VoIP call

I'm looking for a solution to digitally mix sound clips in a SIP or IAX VoIP conversation. Ideally I want a solution on Linux platform. My end game is to have a sound board to play funny clips while ...
0
votes
0answers
13 views

No audio after SIP reinvite

I am trying to get a Gigaset C470 IP to work on my Internet connection. I have figured out the correct configuration for the device and my router (a pfSense sitting between the DSL modem and the LAN) ...
0
votes
0answers
5 views

VoipMonitor - how to limit number of MOS calculations

VoipMonitor begins to calculate a given call's MOS score right after the call ends. If there are many calls that drop at the same time, VoipMonitor will spin up many threads to calculate the MOS ...
0
votes
0answers
16 views

IPTables and SIP Proxy-Authorization Header

Is it possible for iptables to check SIP packets for Proxy-Authorization headers, and if there's more than one header in a SIP packet, modify the packet so only the latest Proxy-Authorization header ...
0
votes
0answers
21 views

set Cisco SPI firewall to high, but allow SIP traffic?

How can I set the SPI firewall settings to "high" for the router, and yet allow SIP traffic? If I set the firewall to medium, the call will connect but I don't hear anything.
0
votes
0answers
53 views

ekiga — could not register (Forbidden, please check that the username and password are correct)

What are the connection parameters for ekiga? I went to ekiga.net and reset my password, still no luck. I've tried more than a few variants: Add a SIP account To add a SIP account, simply click ...
0
votes
0answers
17 views

Twilio Android client: echoing voice?

I am using the Twilioid app (https://play.google.com/store/apps/details?id=com.nethram.twilioid&hl=en) for my android phone and the person receiving the call reports that they can hear their own ...
0
votes
0answers
25 views

how much data in KBytes does facebook consumes per minute

I want to know which codec is used by Facebooc VoIP call. also I want to know how much data is consumed per minute on a facebook VoIP call. I assume it is G.729
0
votes
0answers
34 views

My voice on skype skips/cuts outsooner or later, recurring

I have posted this before on multiple sites, I'm hoping another go may lead to better results. Luckily it more or less stopped five months ago, now sadly it has started again. As the title says, my ...
0
votes
0answers
29 views

Secure Free VoIP Line for sending faxes from home using an HP OfficeJet 4622?

I have an HP OfficeJet 4622 at home, but I do not possess a lan line for sending faxes. I read that it's possible to use a VoIP line for sending faxes provided you use very specific settings. ...
0
votes
0answers
211 views

Assistance needed install UCT IPTV AS (UCT Advanced IPtv)

My IMS system have 6 component : P-CSCF S-CSCF I-CSCF HSS Media Broadcast Server UCT IPTV (UCT Advanced IPtv) as server Each component run in each IP. But I can’t install the last component UCT ...
0
votes
0answers
45 views

Digium card shows green and red light after each second after configuration

My digium card and pri was working fine in other system but i removed and installed that digium card in other system .I have configured it but its showing only 24 channels but i have 30 channels .My ...
0
votes
0answers
30 views

No voice through anyconnect with android phone

I have Cisco CUCM 10.5.1. I connect to it via cisco anyconnect client and register normal using the third party client (droid) on android phone through 3G mobile link. I try to call through the h323 ...
0
votes
0answers
10 views

Make a SIP address call

I have a Polycom Deskphone (model 321) that uses SIP/VOIP to make calls etc via my provider. Calling normal numbers works fine as they are then translated by my provider & routed accordingly to ...
0
votes
0answers
44 views

School Bell with Digium Switchvox

Wondering if anyone has a script or suggestion on how to setup a School Bell with VoiP phones. Basically I'd like to setup a schedule to call every phone in a group, switch to speaker, and play a ...
0
votes
0answers
98 views

Any way to hook up Google Talk with a FreeSWITCH server to make outbound phone calls?

Once, with Google Voice used XMPP, you could use it to do free outbound US and Canada calls with your Asterisk or FreeSWITCH server. Today, if you use this guide, ...
0
votes
0answers
449 views

How to call an extension directly from outside with Asterisk?

How can I call an extension from outside directly without using a voice menu? For example I am calling a number xxx-xxxxxxx and the extension is 2002. I have not setup an voicemenu type thing with ...
0
votes
0answers
859 views

Android SIP / VoIP with doorbell: View video before answering

I am currently using the 2N Helios IP Vario doorbell device. It can be used via SIP protocol, but can also communicate with the 2N Android app directly, without an additional server, which I would ...
0
votes
0answers
543 views

Android and Windows (or Linux) software for simple P2P VoIP

I'm looking for a simple way to set up a VoIP connection between an Android phone and a Windows or Linux computer. I've been looking at Asterisk and other SIP servers, but a) they seem primarily ...
0
votes
0answers
117 views

automated text to speech phone call from ubuntu server to deliver one time confirmation code

Basically, I want to implement the same type of system Google and other sites use to take someone's phone number, and call the number from the ubuntu server with an automated voice that presents them ...
0
votes
0answers
781 views

Yum hangs on setting up package sacks

I'm having trouble with Yum installing postfix. I'm using yum on a trixbox/freepbx system and trying to get voicemail-to-email setup for VOIP Cisco phones in an office environment. Anyways I try ...
0
votes
0answers
836 views

Converting G.722 audio files using ffmpeg

I have a G.722 RTP stream that was captured with Wireshark, and am trying to convert it to PCM using ffmpeg. I have done similar things with other codecs (like G.729), and the conversion works ...
0
votes
0answers
375 views

X-Lite v4 softphone doesn't show up onscreen in Windows 7 (only nonfunctional tray icon)

I installed X-Lite on a 64-bit Win 7 machine. Install seems to go smoothly, then when I launch the application, its splash screen appears, then it goes away, leaving a tray icon with the familiar ...
0
votes
0answers
146 views

encrypted voip traffic for voip provider that doesn't offer encryption

I currently use a VoIP service with one VoIP provider. However, this company doesn't offer encryption to VoIP traffic. This mean that all voice transmit via my IPphone can be sniffed and listen. (I ...
0
votes
0answers
827 views

How to Change SIP default port 5060

Port 5060 is blocked in our network, and I was wondering if it was possible to change it to 5222, which is a port used by Google Talk. If it is possible and I do change it, will I still be able to use ...