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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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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413 views

internet traffic slow through switch; VOIP issues too

There's already a post about this, but the solution to that one didn't work for me, so here's my version: We have Cox cable internet, 100mbps | We have a Sonicwall TZ 215w | We have a netgear L2 ...
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58 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 ...
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327 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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33 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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14 views

SIP server name Defaults to $VERSION for Cisco SPA 942 hard phone

In reading the fine manual for a Cisco SPA 942 IP phone, branded as Linksys: what is, in particular, meant by: SIP Server Name Server header used in responses to inbound responses. Defaults to $...
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54 views

linphone: remove/hide “default identity”

Linphone automatically generates a computer-local SIP identity called "Default Identity". "Real" SIP accounts you have at a VoIP provider need to be created as "Proxy accounts": The main window ...
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179 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 ...
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740 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) ...
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334 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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182 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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1k 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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771 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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286 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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56 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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1k 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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306 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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176 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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172 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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983 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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388 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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15 views

How is a SIP invite sent?

The telnyx Data API tab describes a SIP request: Requires IP Authentication to be setup through the portal and associated with LRN under Telephone Data Tab Send a SIP Invite to lrnlookup....
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4 views

WatchGuard SNMP Traffic Accounting vs ISP Accounting

Our Setup I currently have a WatchGuard XTM 2 Series running as our firewall. There are two Internet connection lines: one EFM line for data traffic and a high-speed ADSL line for VoIP traffic. The ...
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23 views

Recovering Google Hangouts call history

When a user changes their Google account password, they are automatically logged out of Google Hangouts on any and all Android devices. This has the unexpected consequence of deleting all their call ...
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14 views

AT&T DSL causing static on VOIP line

We recently changed from Comcast internet to AT&T DSL. Since then we're having static on our phone line, which is a Vonage VOIP line. I'm assuming I need a filter, but I'm not sure where to put it ...
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25 views

Is it possible to avoid long distance charges with an IP-based phone?

Is it possible to avoid long distance charges with an IP-based phone? I am not talking about client-based VOIP services here (like Skype), I am talking about a REAL IP phone using Asterisk or some ...
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77 views

How to call other phone in the same house and network

I have an analog phone (Panasonic KX-TS505MX) and it is connected to my modem router (ZTE F660) via POTS. My router modem have 2 POTS slot, my question is if I buy another phone, is it possible to ...
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22 views

Belkin Router OpenWRT Custom iiNet Model

I am trying to install OpenWRT or DD-WRT on an old belkin ADSL/VoIP router/modem. But, because it is a custom model made for iiNet the model number isn't listed anywhere. I have opened up the case ...
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35 views

Why is my sip software calling 105

I have a very bad sip provider "vbuzzer" but there prices are cheap ,they don't have an app for android (there is one in the store but i am pretty sure it is not official ,and a scam) and their ...
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43 views

Routing VoIP traffic across VPN?

Context (the Why): I have an RV (recreational vehicle) that I travel around in, connecting to various campground wifi APs (and sometimes my own Verizon Mifi hotspot). I have a VoIP phone I use for ...
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52 views

Linksys SPA3102: receive PSTN calls on sipphone, route outgoing calls to voip provider

I'm having problems to configure my Linksys SPA3102 according to my needs. I have a sipdiscount.com account, a PSTN line, a SPA3102 router (just bought it) and a sip phone. Until now I used my sip ...
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22 views

SPA504 SPA525 2nd Line Ringing when using first Line

Been fighting my provider for 8 months on this. Have SPA504 and SPA525 Phones. When on a call on line 1, when a 2nd call comes in, instead of ringing line 2, line 1 rings again. It flashes indicating ...
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12 views

Analog Centrex to Digital Centrex

Is there a way by which we can convert analog centrex to digital centrex? I have analog centrex at office and we want to convert it to digital so want receive and call using PC. Is there a way we can ...
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30 views

Cisco IP Phone drops call when using group pickup

I have a cisco ip phone that is part of a group. When a call comes through, one phone rings for 3 rings and then rings the group (I believe this might be called a "Hunt Group"). If I group pickup ...
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121 views

Can I use a Vonage ATA as a Google Voice device?

I have a never-used Linksys WRT54GP2 Wireless Router with Vonage enabled phone ports. Recently, I heard from a friend about his successful use of Google Voice with an Obi100 device, replacing his ...
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37 views

VoIP call interruption with my internet

I Have 2 D link 2025u Dsl Routers with 6Mbps Download speed and 0.60 mbps upload speed and i have total 10 user to call over internet via skype and voip connect software, We facing problem with call ...
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49 views

Program won't finish loading until I lock and unlock workstation

We have a unified communications app (Mitel MiCollab UCA 6.0) that freezes on the splash screen for only one computer. We have dozens of users with this app installed, on similar machines with the ...
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31 views

Setting up landline on mobile using Asterisk

I recently got a new broadband connection with a dlink DSL-2730U N150 wifi router. I also got a landline connection through the same line connected to my landline phone. I read at places the I can ...
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75 views

Asterisk server asks peer on other server to Authenticate

I am trying to have an SIP peer on one asterisk server call an SIP peer on another asterisk server. I have SRTP enabled between the two servers not the SIP clients. When I initiate the call, i.e SIP1 ...
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247 views

How do I connect my D-Link DIR-605L wireless router to my TP-Link TL-SG2424P PoE switch?

We connect to the Internet via our D-Link DIR-605L wireless router. We just bought a TP-Link TL-SG2424P PoE switch to power our new VoIP phones and connect them together. (Our wireless router will ...
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457 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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122 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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746 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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1k 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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191 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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1k 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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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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532 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 up-...
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162 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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1k 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 ...