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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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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9 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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2answers
45 views

Jitsi and Telekom Germany - outgoing calls “Call failed - Not Acceptable”

I am trying to configure Jitsi to operate as a softphone for my Telekom Germany VoIP-connection. So, I went to the website of Telekom Germany, which provides the following details for connecting: ...
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22 views

What port does the DOCSIS 3.0 update come through on a firewall?

The problem is my cable company tries to push updates to a cable modem that they don't own which causes home network problems. Is there a way to block their DOCSIS updates, but not the ones ...
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1answer
18 views

Do FreePbx need VPN for connection?

I heard about that PBX (like freepbx, elastix, etc...) need a VPN to users connect to it. if is it true, why it needs? if so, how can I do not use it in the user client? any alternatives? i heard ...
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13 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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1answer
52 views

Failed to authenticate SIP peer

I am trying to connect an SIP peer using Zoiper to my asterisk server. The peer is a soft-phone on my server. I have the following config for the peer: [201] disallow=all allow=alaw ...
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1answer
34 views

What's Lax and Lax2 protocols?

Are LAX and LAX2 communication protocols like SIP? What are these? Some telephone systems support them beside SIP but I cannot find information on google
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2answers
21 views

Why do Ekiga touchtones only work with the RFC2833 setting?

I have Ekiga acting as a softphone for my VOIP account. I made a call to an automated switchboard which required me to "press 2 to continue". I attempted to press "2" on the keypad pane but the DTMF ...
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1answer
50 views

Does AT&T U-Verse phone service cause problems with modems?

I have landline phone service with ATT Uverse. My router is a 2wire, and when I go to the router homepage, I see something that says the VoIp service is either idle or active. I also have a modem in ...
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29 views

Is there an easy way to reduce background noise in skype (such as through a voice activation detection meter)?

Is such a thing built in to skype or has a program been made so that your mic only activates once the sensor detects past a certain threshold? I know it is common in almost every other VOIP service ...
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1answer
36 views

Connect LAN-device to iMac socket and have internet (through WiFi) on LAN-device

I have an iMac which has internet through WiFi. I also have a VOIP-phone with only a LAN socket (no WiFi adapter availalble). I would like to connect my VOIP-phone to the LAN outlet on my Imac, and ...
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65 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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1answer
23 views

server with multi purpose possible?

I would like to have the following 3 uses available to me. Accessing my files within and outside my network Game servers occasionally; Minecraft or Gmod (as example) Teamspeak 3 Is it possible to ...
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89 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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24 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 ...
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56 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) ...
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1answer
61 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 ...
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1answer
90 views

Incoming packets drops when a lot of packets

I encountered a problem with my network. When I play online games that send many packets, my ts3 connection statistics show 25% - 50% packet loss. First game I noticed to do so is Space Engineers, ...
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11 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 ...
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27 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 ...
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39 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.
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41 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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109 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 ...
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1answer
63 views

topology for an unmanaged switch (firewall)

I'm considering an unmanaged switch. My understanding is that many of these devices explicitly lack a firewall. In that case, what do you use as a firewall? I suppose one option would be to use a ...
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61 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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1answer
14 views

VoIP Phone Reboot Leads to Internet Access? [closed]

So this isn't really an issue about not having access or anything but rather as to why this issue was resolved with the method I used. So in the morning, I tried to connect to the internet but I ...
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1answer
425 views

VOIP Fax + Raspberry Pi

I just moved into a new office and don't want to get a landline phone. I'd kind of like to set up an Asterisk box and use Google Voice as my service provider. The main things I'd like to have are: ...
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26 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 ...
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1answer
124 views

Separate VoIP traffic from Data traffic on a computer that has a softphone

I have been entrusted recently with a task that seems a bit more complicated than I initially thought. I would have asked Google, but I can't quite formulate the right question using just a few ...
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1answer
104 views

Asterisk Ivr menu mysql build

I would like to implement an IVR menu to see if the call comes in and announces a text waiting 6 numbers. Then the number 6 is controlled by a MySQL database. If "Active" or "Valid" the status will be ...
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1answer
125 views

How to “hide” VoIP traffic?

I'm pretty soon gonna be punished by my telecom operator for using WhatsApp,Tango,Skype etc. From now on not only will I be charged for the 3G Connection that makes services like WhatsApp,Skype etc ...
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387 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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1answer
187 views

Plivo SIP problems with NAT traversal

Ok, I've looked and looked, and can't seem to find anyone that's written about this problem. I have a simple office setup where there are two SIP phone extensions (Polycom 331) behind a Sonicwall ...
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1answer
124 views

Can Asterisk accept and send SMS?

Is it possible to send and receive SMS over Asterisk? I am using Asterisk for my communications and it works pretty good. I just wish that I could send SMS to my callers. thanks
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32 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
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1answer
17 views

Discovering the VoIP connection parameters for CISCO 7961

Not sure this question belong here, please redirect me to somewhere else in case. I've a CISCO 7961, a VoIP phone, which receive its configuration from a company central server. I asked the IT people ...
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1answer
1k views

How to configure linphone for video calls?

I’m trying to enable communications between the following clients: Nortel IP phone 1535. Linphone 3.7.0 running on FreeBSD. Linphone 3.7.0 running on iPhone (Linphone iPhone 2.2.3). The good part ...
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42 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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32 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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1answer
274 views

Freepbx outbound calls through an FXO device (Grandstream GXW4108)

Im trying to use GXW4018 fxo gateway with freepbx. Inbound calls work as expected but outbound calls doesnt. Outbound calls work but i cant seem to be able to force an extension to dial out with A ...
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240 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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1answer
750 views

Android won't register SIP on FritzBox router

In the past I've used VoIP/SIP apps to call from my Android phone. Recently I got the upgrade to Android KitKat (4.4.x) and I've noticed that it includes SIP accounts natively (accessible through ...
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1answer
378 views

Reboot or restart Polycom SoundPoint IP 335 from command line

I have 4 Polycom SoundPoint IP 335 IP phones. Just wondering if anyone knows if these phones can be restarted/rebooted from a command line. They all have static IP addresses. I do have access to ...
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175 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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59 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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37 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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1answer
754 views

Cisco VoIP phone - Strange behavior after router reset

We're an IT company based in London, but one of our clients is in New York. They have a Cisco SPA504G VoIP phone connected via a set of net gear power lines, and then to their consumer router ...
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1answer
26 views

Where or how do I register a new phone number intended to be used with voip? [closed]

Ok I'm not sure if this is the right place to ask. Goal: Have a voicemailbox with multiple phone numbers. 1 in each continent. Example: +1123456 -> voicemailbox +49123456 -> voicemailbox The scope ...
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16 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 ...