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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3 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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29 views

Realted to VOIP Asterisk DID and Toll free number [on hold]

i am new to this forum as a register user. but i am not a new user in this forum. from year 2004 onwards i am getting all help from this site. this question answer are so meaningful, i was doing all ...
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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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6 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 ...
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9 views

Single Number call routing (most preferably GSM) [closed]

I have a specially bought GSM SIM card (for special number). I want to setup a call center with 6 PC's which will receive calls from that SIM card number in parallel. How can I do this? Some one ...
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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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11 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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15 views

Using a Barracuda F400 Firewall Router Without a Modem [migrated]

I'm looking into a problem where VoIP phones don't ring for some of the incoming calls that I believe might be due to the network configuration. Do you need a modem if you are using a Barracuda F400 ...
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1answer
34 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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42 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
38 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
19 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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28 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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0answers
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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1answer
85 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
21 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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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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0answers
48 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
126 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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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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1answer
384 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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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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1answer
79 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 ...
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1answer
91 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 ...
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1answer
228 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
123 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
46 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. ...
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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 ...
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1answer
67 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
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1answer
150 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 ...
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1answer
275 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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109 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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2answers
2k 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 ...
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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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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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3answers
66 views

Does DSL/VOIP Service Work In A Power Outage

Just curious, but does DSL service work in a power outage? From what I know, DSL uses regular phone lines to transmit digital rather than analog signals at a higher speed, and regular phone service ...
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2answers
147 views

Difference between VOIP and IP Telephony

I am studying CCNA, and in the curriculum I found something like following... ↓↓↓ VoIP When introducing VoIP to a network that uses traditional telephones, it is important to remember that ...
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1answer
510 views

Router using different ip address when modem is in bridged mode

I have a zyxel modem/router that is blocking my voip, and so I want to put it into bridge mode. When in bridge mode, I use my own router, and my voip works fine. We have 5 static ip addresses at our ...
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2answers
85 views

What happens when two SIP clients log in at the same time

If two SIP clients are logged in to a service provider at the same, what happens when an inbound call comes and what happens if you try to make an outbound call? (This question is regarding SIP ...
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1answer
87 views

The Risk of Port Forwarding to a VOIP Device

My VOIP provider uses a device to supply the service. The device is on my side of my router and part of my LAN. He requires I forward a very large range of ports to his device (something like 5000 ...
2
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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
335 views

FreePBX Calls connect but no voice

I have FreePBX running on virtualbox on a windows 8 host the network adapter is set to bridged and I have assigned it a static IP address. My windows 8 host is connected to a typical home wireless ...
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0answers
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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100 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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1answer
45 views

Disabling saying extensions when the ext is not available with Asterisk?

I setup Asterisk and everything works great. The main issue I have is that when the external call cannot reach the default extension, the system repeats the extension number which could be desirable ...