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16

Yes, there is a software/hardware that can achieve this, and relatively cheaply. Asterisk is an open-source (free) PBX software, that can do what you need, and more. Asterisk is usually used on linux, although there are pre-packaged virtual machine appliances available. There are also lots of Asterisk-derivatives, like PbxInAFlash, TrixBox, and others, that ...


13

http://xmpp.org/ aka jabber


9

Take a look at Empathy, it's a multi-protocol instant messaging app that, unlike Pidgin, does support Google Talk voice chat. It's available in the Ubuntu repositories. You'll also need to install a backend for empathy in order to use gtalk. Install both with: sudo apt-get install empathy telepathy-gabble


6

Skype is a type of VoIP, so a better differentiation is whether to go for a Skype-based solution or use something like an Open Source Asterisk or commercial VoIP system. For general one-to-one calling with a few other telephony features, Skype will do the trick, but if you want traditional PBX-type features such as: Inter-site transfers by extension ...


6

If you use Chrome as your browser, then you can get a Google Voice account and the Google Voice extension. Then you'll be able to click on a number and it will route it to your phone.


6

Yes, mobile phones (mainly 3G/4G phones) use TCP/IP for transferring data (such as browsing the web, downloading apps, etc.) As for the phone calls, those don't really use VoIP, but are more a function of the technology the carrier uses (Verizon uses CDMA (4G is LTE) and AT&T uses GSM).


5

The best is to block all the ports (iptables default to DROP) and open only what you need (probably port 5060 at least for your asterisk server). If you need to administer asterisk from outside, port 80 needs to be open if you administer it from a web browser or port 22 if you use ssh. Update: For Asterisk, I don't think you need other ports open and even ...


5

So far I have only had success for text only support using Pidgin with the SIPE plugin. I'm looking into SIP Communicator (which is java based and runs on all platforms) it does have some VOIP and Video support, but I have not been able to get it to connect to my company's OCS SIP server. For Pidgin, get at least version 2.6.6 and install Pidgin-SIPE plugin ...


5

You can do it with Sipgate and an asterisk server. I set mine up like this a few weeks ago. There's a pretty comprehensive guide and lots of followup posts here. http://forums.slickdeals.net/showthread.php?sduid=0&t=1480557 There was actually a followup thread on SD where people are running this setup without a local asterisk server, and using a ...


5

I think GTalk works on linux... Look at Tapioca Look at the Telepathy client for gnome. Here is a link to the GTalk client reference page. And the Pidgin FAQ. Try Pidgin first.


5

Linphone should do the trick. It supports call encryption via ZRTP. NAT traversal is easy with it via specifying a STUN server (i.e. stun.ekiga.net). It's super easy to set up. However, as it is SIP based, you will need a SIP server to go through. Personally, I just use sip.linphone.org as my proxy. However, you should be able to set up SIP Witch on both ...


4

Normally, routers have a page called QoS (Quality of Service) which allows you to do exactly what you are asking for. The configuration page varies from router to router. In my Cisco router it is: . You can pick an application, or define a new one, and then assign it a varying priority. Since I do not know make and model of yours, I cannot be more specific, ...


4

Skype should work fine How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers ...


4

No numbers are inherently "SIP" numbers or any other type of numbers, they are just numbers. Numbers are handed out to VoIP providers and PSTN line providers alike from number vendors (such as 360 or Level 3, they do not generate the numbers themselves - though they can, however this doesn't mean they own the number, it will route only within their network ...


4

From personal experience, I prefer Google Voice. The service will allow you to port over your existing phone number to Google. The features it has are awesome: voicemail transcribing, massive contact support, SMS, and so many more I can't list. The awesome part is that you can make Google Voice ring multiple phones. It could ring a cell and a home line, or ...


4

Under Windows, you can create a new protocol using the Registry Editor: Start / Run / regedit Look for the HKEY_CLASSES_ROOT hive, and create a new series of keys with the following structure: tel\shell\open\command At the tel level, edit the (Default) string, and change its value to URL:tel Protocol Again at the tel level, add a new string called ...


4

If you are in the same situation, a solution I came up with is to point the registry command for the protocol to a batch file that evaluate the command line parameter stripping the "tel:" part and start QuteCom... So here is what I put as string in the registry command key: "C:\telProtocolHandler.bat" %1 Here is the content of my telProtocolHandler.bat ...


4

Everything will be a little slower over the VPN, since the connection naturally introduces several more hops to and from a remote location. For example, if you live in London, the VPN server is in USA, and you are connecting to another London-located server, then the packets will have to go overseas and back. Depending on the connection quality, the delay ...


4

Actually VoIP is not a protocol but an umbrella term to cover voice communication over IP. So SIP is considered to be VoIP, and so is Skype. So it's called a SIP phone since it's using the SIP protocol but wouldn't work with any other VoIP protocol. Perhaps you were referring to the H.323 protocol? If so, the differences and similarities between H.323 and ...


3

I am pretty sure these companies use ordinary SMS gateways, such as Kannel. My company is provided with this service by Beepsend and we can send SMS messages through a SOAP web service and specify any caller ID we want (alphanumericals allowed). Of course, using these services to send from a caller ID that belongs to someone else without their permission is ...


3

you have to bridge the two connections though. in windows, if you go the the network connections control panel, select both of the connections, then right click and choose "bridge connections" you can then plug another system(your VOIP for example Dev) into the laptop, and then be able to access the network across the server.


3

Linux command line: linphone includes a scriptable linphonec command-line version. Starting linphonec with the --pipe option will create a socket in /tmp that one can write to and read from. It fulfills all your requirements and I've tested (for my own uses) all of them successfully: Calling via SIP, transmitting DTMF sequences, recording a call to file, ...


3

BillP3rd's answer will help you with setting up QoS, but this will not solve your call quality problem. QoS will indeed prioritize your traffic, but only up until it hits the gateway. Once it leaves the router and heads towards the first hop, nothing is prioritized any longer. As far as your ISP is concerned, it is all standard data traffic and any tagging ...


3

Give Mikey's guide on 2nd LAN subnets a miss. Take out the firewall filter, disable 'for IP routing usage' (2nd subnet) and have a read of this guide on address mapping instead: http://www.draytek.com/user/SupportFAQDetail.php?ID=1353 Incidentally, I have never had an issue connecting a SIP phone via NAT, provided that you setup the right port forwarding ...


3

The stock figures for a VoIP connection using a plain vanilla G.711 CODEC and well-compressed G.729 are: G.711 is approx 76.6MB per hour full duplex G.729 is approx 27.4MB per hour full duplex Which, I am told, means using your VoIP phone with a G.711 CODEC for one hour a day is about 2.3GB per month. More info at: http://www.dslreports.com/faq/12190


3

Yes. To place a call with Google Voice from a regular phone line, you simply dial your own Google Voice number from any regular phone, press 2 to place a call, and enter the number you'd like to call. Here's more information on this: http://www.google.com/support/voice/bin/answer.py?hl=en&answer=115079


3

While it's possible, it's not likely. There are a few simple ways you can tell for sure. First, do a traceroute from one client to the next to make surethe network is configured properly. If traceroute packets are leaving the LAN, somethig is seriously misconfigured. Next, start a VoIP call. After the call has started, physically pull the plug on your WAN ...


3

What you want to do is set up a SIP server on your LAN. There are several, then you'll be able to use SIP clients like Ekiga, or whatever the windows one is called, to get Skype-like (VOIP) functionality. There are loads of such servers, like asterisk or freeswitch.


3

IRC will do all your 'chat' needs. There are several options for servers and clients (even web based). Just wouldn't fill your 'video' optional request.



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