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.

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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 ...
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1answer
4k 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: ...
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1answer
183 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 ...
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1answer
977 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 ...
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1answer
65 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 ...
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1answer
111 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 ...
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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 ...
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284 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 ...
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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. ...
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49 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 ...
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131 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 ...
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381 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 ...
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165 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 ...
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48 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 ...
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569 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 ...
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197 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 ...
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159 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 ...
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163 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. ...
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659 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 ...
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347 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 ...
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9 views

Cisco CUCME Outbound Calls from IP phone to PTSN

I do not expect an outright explanation, but I am wondering at least where to look for the answer. I think it may be in CVoice. It is not explained on the CCNA level. Incoming: PLAR to IP Phone ...
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11 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 ...
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12 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. ...
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45 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 ...
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29 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 ...
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12 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 ...
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22 views

Unable to set did number as caller number

Earlier i was using one digium card in a system with 30 channels and 100 did numbers( 5059500 to 5059599) given by my provider(say provider1) on pri line.Everything worked fine means i can set the ...
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14 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 ...
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8 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 ...
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28 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 ...
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63 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, ...
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110 views

3CX integration with SQL server database

I am planning to buy 3CX for my company. I have my own call center software which is used to enter the details about caller. When our customer care executive enters the caller number in the grid, ...
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192 views

FreePBX, no IP Phone connected: queue not calling external numbers

I run asterisk/FreePBX installed with the FreePBX distro on a dedicated server. There is Queue for incoming calls with 2 static agents: extension 01 and 02. Each static agent has a VoIP phone ...
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113 views

Polycom IP 501 - No web interface?

Just my luck. I ordered a polycom IP 501 phone off the internet and can't get it to connect to my voip service (yes, my settings are correct). So I try to go to the web interface and get a 404 - page ...
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88 views

How do I setup a Yealink phone over internet connection sharing?

The physical configuration is as follows: Computer accesses the internet over 'Wireless Network Connection' Yealink Phone is connected to the Computer over 'Local Area Connection' Here are the ...
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101 views

How to use SIP soft phones to test concurrent calls?

I don't know anything about softphones, VOIP, SIP, etc, but I want to setup a development computer to dial into a SIP server (twilio app) with concurrent calls, using some kind of softphone software. ...
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354 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 ...
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169 views

Passing through TOS/DSCP through two layers of OpenVPN, possible?

About a week ago I posted about a problem I am having getting usable VoIP connections working on a machine configured using mirimir's dual-VPN isolation system. (For details on the setup see my ...
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16 views

setting up phone recording system

I would like to have a system (either hosted on our own computer or provided by some sort of service) where we can receive calls to a virtual phone number and allow users to record messages (sort of ...
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377 views

FreeSwitch - how to configure ICE, Turn, Stun, NAT traverse mode?

I have been using FreeSWITCH but some of my test users are behind very complicated network. Where default FreeSWITCH setup is not working such as: they can't register via SIP and also there are audio ...
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685 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 ...
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414 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 ...
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96 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 ...
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0answers
613 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 ...
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207 views

Firefox can't find the file at sip: xxxx

Hi have a voip service that works great and use Bria Counterpath softphone - also good. I have installed a click to call add on for Firefox and keep getting this error: Firefox can't find the file ...
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687 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 ...
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339 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 ...
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144 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 ...
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758 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 ...
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8 views

Can't register account from Zoiper to Elastix

I have Elastix 3.0 stable installed in Virtual Box. I can access the administrator panel from my computer, so networking part between host and virtual machine is fine. I added an extention 101 ...