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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32 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
32 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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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 ...
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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 ...
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20 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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20 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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49 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
49 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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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 ...
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1answer
13 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
205 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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16 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
67 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 ...
0
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1answer
60 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
101 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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222 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
124 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
86 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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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
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1answer
13 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
696 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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32 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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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. ...
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1answer
206 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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201 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
443 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
173 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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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 ...
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42 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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29 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
496 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
22 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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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 ...
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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 ...
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1answer
133 views

How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network

    I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want ...
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42 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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1answer
1k views

Voip phones losing registration from external SIP PBX

This has been a pain. My polycom voip phones (about 10 of them) are all losing registration every few hours. Each time I check them, some will be registered and some will not. Rebooting my firewall ...
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96 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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1answer
117 views

Why a DNS server on an ATA?

I was portscanning this home network from the outside and found three open ports: telnet(!), http and domain. I quickly realized that those ports were belonging to an Eltek R7121-L1 ATA that is ...
0
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1answer
145 views

Creating an outbound call center software in asterisk [closed]

I an outbound call centre as i know the server calls a person and connect the call to the executive.Executive via headphone attached to a system directly talk to the person means there is only one ...
0
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1answer
333 views

Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I ...
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1answer
198 views

Identifier 2, identifier_type 2 not found in identifier list

I am using asterisk 11.9.0 and created a dialplan.My problem is that everytiime i perform an sql operation its working fine but the line same => n,mysql(clear ${resultid}) is giving a error ...
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1answer
52 views

Why use a PCI System Timer?

I have heard it said that servers serving VOIP services such as Asterisk should use a PCI System Timer. Why? Nowhere is this ever explained, and googling reveals no clear use-cases for the device. ...
0
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1answer
13 views

Can you automate change of settings for all phones?

Is it possible to automate the process of changing some settings for VOIP phones running on Asterisk PBX? i.e. I have 10 phones connecting to PBX at all time. 5 of those phones are at one customer ...
0
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1answer
98 views

Why isn't my router allowing my PAP2T to connect to my VoIP service?

My tech knowledge is middle-low: as you can see I'm not shy about getting into the interfaces for my router, but have no idea what the more complex settings mean. I got a new wireless router (D-Link ...
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20 views

Using an existing telephone system with Skype home service [duplicate]

I'm interested in migrating my current telco-based phone service to a more affordable VOIP solution. Because of this, I've decided to settle on Skype, which I've used for years and has been my choice ...
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3answers
2k views

Skype shows I'm online when I don't have any Skype instance running

Recently Skype shows me as online to my contacts even when I do not have any instance of Skype running on my desktop I signed out of Skype on my mobile Apparently, this is a feature not a bug. ...
2
votes
1answer
205 views

Why my outgoing VoIP calls are sometimes that bad?

I have a Cisco SPA122 ATA Router in Bridge mode where I connect my analog phone. Turns out that usually on incoming calls, I don't have any issues: connection goes smoothly, the caller hears me ...
0
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1answer
344 views

Websockets on Kamailio + sipml5 + rtpproxy-ng + jitsi

I need quick help with pointing direction on what I should look at - I got Jitsi with Kamailio up and working both signaling and RTP streaming now I got Kamailio with ws:// setuped and sipml5 working ...
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2answers
4k views

Skype cutting out and skipping my voice

In Skype calls, sooner or later my voice starts cutting in and out; mid sentence my voice will go silent for several seconds, with no quality loss outside of the cut out. Sometimes instead my voice ...