VOIP is an acronym for Voice Over IP, or in more common terms phone service over the Internet.
If you have a reasonable quality Internet connection you can get your phone service delivered to you through your Internet connection instead of from your local phone company.
 
  Some people use VOIP in addition to their traditional phone service, since VOIP service providers usually offers lower rates than the traditional phone companies, but sometimes don't offer full 911 service, phone directory listings, 411 service, etc.  

You can call the following VoIP numbers for free. From a softphone, just copy&paste the full SIP address (sip:user@domain.com). From a hardware SIP adapter, copy&paste the full SIP address into your speed dial web page. All of these numbers worked in September 2005.

BC Wireless autoattendant sip:16044845289@sip.like2fone.com
Caller ID readback sip:16049586111@sip.like2fone.com
Echo test sip:613@fwd.pulver.com
enum2go customer service sip:878107472000010@sip2go.com
ENUM test sip:437203001720@sip2go.com or enum:+437203001720 (no G.729)
Ewing IT sip:611300766674@sip.like2fone.com
FWD Volunteer Welcome Line sip:55555@fwd.pulver.com
Ingate music sip:music@trysip.ingate.com
Intertex autoattendant sip:autoattendant@intertex.se
Lake Bracciano sip:1276282@sipgate.de (no G.729)
MIT sip:16172531000@proxy01.sipphone.com
Party Line sip:12220000000@proxy01.sipphone.com
Personal Telco sip:274185@fwd.pulver.com
Patton Electronics sip:support@patton.com (no G.729)
SIPphone Welcome Recording sip:17474745000@proxy01.sipphone.com
SNOM sip:4930398330@sip2go.com or enum:+4930398330 (only answered during German business hours)
TELL sip:18005558355@proxy01.sipphone.com (news, sports, weather)
UCLA sip:13108254321@ucla.edu (no G.729)
U. Philippines sip:0116329818500@proxy01.sipphone.com
VoipTalk UK Login test sip:902@voiptalk.org
Xmission sip:xmission@pbx.xmission.com (no G.729)

Notes:
(1) To dial from a hardware SIP adapter, you can usually paste the entire address into your phone's speed dial web page. SIP adapters cannot call any IAX numbers.
(2) G.711 is the usual codec that VoIP phones use, but the WiSiP phone works better with G.729-only selected. Calls from a WiSiP phone to VoIP numbers that don't support G.729 will ring and connect, but you won't hear anything after that.
(3) Suggest testing from pulver.communicator or SJPhone softphone. Please list only *open* VoIP phone numbers, meaning that you have successfully made a call to that number from a SIP proxy located in a different domain. For example, you can call sip:xmission@pbx.xmission.com from pulver.communicator which is registered in the FWD.pulver.com domain, so the xmission SIP number is open. On the other hand, Vonage numbers can only be called for free from other Vonage phones, so Vonage is NOT open to calls from other SIP proxies.
(4) If you install SJPhone, it will make SIP a "registered protocol" in Windows, so that clicking on one of the above 'sip:' links will automatically open SJPhone and start the call.

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