Veblungsnes i Rauma, Møre og Romsdal.
Magne Heen <magne.heen@weblung.org>11/01-97Index
Veblungsnes |
Kulturhuset |
Veblungsnes historielag | Raumadalsfolket (Henrik O. Røvig) | Riksvegomlegging
The concept of "free telephony" was first introduced in 1995/1996. The idea is that
your voice rather than traveling over your phone line can travel over your Internet
connection. This is called Internet telephony or: ?Voice over Internet
Protocol/Packet? or VoIP.
Hentet fra: Chat Cord
What is Free IP Call ?
It is an intra community service which facilitates telephone
communication between the members. Members of the network
can communicate between themselves using interactive voice
chat.
Free IP Call / Global Network (FIPC/GN) is different from
traditional IP telephony services. This time you can do
WITHOUT your computer and use your usual telephone headset !
Being a member of FIPC/GN, you can join another user of the
system with a personal identity number.
Private Number Services, Enum Services, Network Peering and Exchange, Peer to Peer Service, Free PSTN to VOIP Gateways, Asterisk to/from PSTN services, VOIP to/from PSTN services, Big Guys & Back-End Providers.
Please complete this brief form so that we can email your free calling card to you and provide you with free membership in PhoneHog.com. As a member of PhoneHog.com, you will receive many exciting opportunities to earn free long distance calls on your new calling card. Thank you.
Ser ut til at du blir avbrutt med reklame i samtalen. Sitat: "Let your ADVERTISERS pay for your long distance calls..)"
Website: Website: http://www.freeworlddialup.com/Quoting from website:
Free World Dialup (FWD) allows you to make free phone calls over the Internet using a 'regular' telephone or a computer program. The
FWD service has over 50,000 subscribers from over 150 countries. You can join today, at no cost, by either using programs already on your
computer, or by downloading a free program. Later, an IP phone upgrade will let you realize the total freedom FWD provides whether on
Cable, DSL, Dialup modem, or WiFi around town.
FWD is based on SIP Experimental IAX support is also availible, see http://www.fwd.pulver.com/advanced/iax
Troll Phone Networks AS is a privately
held company based in Bergen, Norway.
We provide services and solutions based on
Open Source software, in particular Linux
and Asterisk PBX.
We expect to be registered as a phone
operator in Norway late 2003 or early 2004.
Once this has been done, we will start
delivering phone service over broadband
network connections.
SIP, the Session Intitation Protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services. Abstract from the SIP RFC 3261 This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols. SIP is very much like HTTP?, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.
Community
Skype is all about the community of people using it. Our family grows with each new user coming to Skype, and we’d like your help in sharing the message. In return, we offer you cool backgrounds, Skype-Me buttons and e-cards and give you a
chance to discuss your Skype experience directly with other Skype users and actual Skype developers.
Vision
Skype is the Global P2P Telephony Company™ that is changing the telecommunications world by offering consumers free,
superior-quality calling worldwide.
“I knew it was over when I downloaded Skype,” Michael Powell, chairman, Federal Communications Commission, explained. “When the inventors of KaZaA are distributing for free a little program that you can use to talk to anybody else, and the quality is fantastic, and it’s free – it’s over. The world will change now inevitably.”
Fortune Magazine, February 16, 2004
“The idea of charging for calls belongs to the last century. Skype software gives people new power to affordably stay in touch with their friends and family by taking advantage of their technology and connectivity investments.”
Niklas Zennström, CEO & Co-founder of Skype
Voop as er en registrert teleoperatør, med avdelinger i Bergen, Oslo og USA. Vi er et 100% norskeid firma som bygger
våre tjenester på moderne IP-teknologi og Open Source software som Linux og Asterisk.
Voop Standard
Bredbåndstelefoni for privatmarkedet til svært gunstige priser. Ingen
bindingstid, du eier selv telefonen eller adapteren (som du kjøper der
det passer deg, f.eks. fra oss) og du kan velge ekstratjenester fra vår rikholdige meny.
Flere tjenester kommer etterhvert.
Abonnement fra NOK 79,- per måned
PC-to-Phone service can be used from any computer connected to internet (minimum connection speed - 28.8 Kb/sec). A full duplex sound card should be installed on the computer from which the service will be used. It is also recommended to use a headphone instead of the speakers and a separate microphone.
Norway, 0.0500 USD per minute
Norway-Mobile 0.3500 USD per minute
Flere leverandører å velge mellom.
Free yourself from your computer with Chat-Cord? and
talk without limitations
Features: Easy Setup. No Drivers. No external power. Works with PC/MAC/Laptop or any VoIP supporting OS. Works with any VOIP, Softphone or Broadband services like MSN, FWD, Skype, Xten,Yahoo, eStara, eyeP, Vonage, Stanaphone, Net2Phone, SkypeOut, TerraCall etc...
You need: PC with Sound Card, Internet Connection, a Telephone (preferably Cordless) and a Chat-Cord?
Connect the Chat-Cord? to the phone-set and to the PC sound card as shown in this picture:
First of all, I must officially advise against connecting anything other to the telephone line than equipment approved for the purpose by the telephone company or some other regulatery body. Telephone company tend to be very strict about unauthorized gear hanging on their lines, and if something does go wrong with your gadget (like putting dangerous voltages to telephone line) you will be in deep trouble.
Back in 1996 I wrote an article for Radio Guide entitled "Phone Line Basics", which attempted to explain audio connectivity through the phone system. At the same time I placed a reprint of the article on our web site. Almost immediately I began to find links into the article from all over the world. After six years it is still the most popular article on our site. One of our competitors even asked if they could place the article on their web site. As it turns out, it is not easy to find accurate information about the phone system available in condensed form.
: Do I need to use a special kind of wire, is a NT1 (also called ntba or network termination device) needed for termination / powersupply
between the card and a phone? A: Yes, unless you have a hfc-s based card that does power the line (I don't think they exist) Just follow the
documentation at http://isdn.jolly.de/ to wire up your card, your NT1 and your phone equipment.
In order to connect a telephone directly to a card, you need to:
This document describes features available with the ISDN Q.931 BRI NT/TE voice modules supported on the
Cisco MC3810 multiservice access concentrator, and voice interface cards on Cisco 2600 and Cisco 3600 series modular
access routers. This module includes the following sections:
The ISDN TA PC Card brings the latest in mobile communication technology
with flexibility and convenience. With the wide range ISDN signaling
protocols switches supports, mobile experts may enjoy the mobile
communication across the world and the high speed digital service.
I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO).I am using zaphfc now. I also use an Epia V 10000 mainboard (with Nemiah processor).
Quoted from http://www.asterisk.org/
Asterisk is a complete PBX in software. It runs on Linux, BSD and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP, H.323 (as both client and gateway), MGCP (call manger only) and SCCP/Skinny (limited). Check the Features section for a more complete list. Asterisk needs no additional hardware for Voice-over-IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks?. In addition, single to quad port analog FXO and FXS cards are available. Other vendors' cards can be used for BRI (ISDN2) or quad- and octo- port BRI based upon CAPI compatible cards or HFC chipset cards.
Which version of Asterisk is documented here?
This Wiki covers the both the stable and the development branch of Asterisk. When adding new commands, applications and options, please also add a note on *when* this was added so that users may compare with their version date.
The Asterisk Manager (am) is an HTML based configuration and management tool designed to work with the Asterisk PBX. It allows you to create dial plans using an easy to use HTML interface, and remotely configure and manage your Asterisk PBX. Administrative functions allow you to start and stop the PBX, create and modify configuration (.conf) files, and view call detail records.
AstWind is a package that allows you to run Asterisk on top of a Windows operating system using Cooperative Linux (coLinux for short). A port of the Linux kernel, coLinux runs cooperatively alongside legacy operating systems on a single PC. It allows one to freely run Linux on Windows 2000/XP, without using a commercial PC virtualization software. With the addition of the WinPCap library, the coLinux kernel has access to the network through the host machine and can fully participate as a separate PC with a unique IP address. A Debian GNU Linux file system, with a full development environment and a pre-built Asterisk installation is provided to get you up and running.
Windowsbrukere trenger denne.
Cooperative Linux is the first working free and open source
method for optimally running Linux on Microsoft Windows
natively. More generally, Cooperative Linux (short-named
coLinux) is a port of the Linux kernel that allows it to run
cooperatively alongside another operating system on a single
machine. For instance, it allows one to freely run Linux on
Windows 2000/XP, without using a commercial PC
virtualization software such as VMware, in a way which is much
more optimal than using any general purpose PC virtualization
software. In its current condition, it allows us to run the
KNOPPIX Japanese Edition on Windows (see Screenshots).
Network support can be achieved for coLinux in two ways:
1.NAT network (Network Address Translation)
2.Native/Bridged network
The better way to connect is by a native/bridged connection to the network since NAT-traversal hinders some
protocols and makes it difficult to run services on coLinux accessable by the network. However, if your
network-provider supports only 1 IP-address, running coLinux with NAT is a way to prevent the setup of a
masquerading router/gateway.
( Note: If you're a home user on a dial-up, DSL, or cable modem connection, you should use NAT configuration
(unless you have a network connection-sharing device). "Bridged" means your coLinux virtual machine on the
"inside" will share the same IP address space as your "outside" connection. Unless you have explicit permission
from your ISP for two simultaneous connections to their backbone and have everything configured properly,
this won't work.)
If you want to obtain an IP-address by DHCP make sure a dhcp-client is installed in advance in your Linux
distribution. For Debian: apt-get install pump
I have 2 HFC-S chipbased Billion Bipac PCI ISDN BRI cards installed in PC. Would like to use one card as in TE and one in NT mode. System works fine running pbx4linux.But want to use SIP functionality, so I would like to try out the Asterisk.
Asterisk has gotten pretty mature since the project started and is nearing the 1.0 release. The only problem is, that the configuration and documentation is scattered around in various places on the internet. There is a draft of version 2 of the handbook, but that was not as fullfilling, as i would have wanted.
Die Kabel-FAQ
Da immer wieder Fragen zum Thema "Wie muss ich ein Kabel fuer xxx loeten?" auftauchen, habe ich hier
diverse Steckerbelegungen, Kabeldaten, Signalbezeichnungen, -spannungen und -stromstaerken sowie spezielle
Schaltungen zusammengestellt.
Soweit moeglich jeweils mit der Bedeutung der Signale, denn es kann fuer spezielle Zwecke noetig sein,
Bruecken einzubauen oder Pins "falsch" zu beschalten, um einen bestimmten Effekt zu erzielen. Das Ganze
traegt weniger den Charakter eines klassischen FAQ mit Frage: Antwort. Das ist aufgrund der Komplexitaet
mancher Sachverhalte nicht zweckmaessig, es schadet nie, etwas rechts und links des eigentlichen Themas zu
finden.
Ergaenzungen und Korrekturen sind willkommen! (an hifi@gmx.de schicken! (Hinweise dazu!)
Hinweis: Die FAQ besteht nur aus einer einzigen Datei, alle Links zeigen nur auf Stellen innehalb dieser Datei.
Wer die Seite speichert, hat automatisch alles beisammen. Eine ASCII-Version (zip) kann man sich hier laden.
NT Kabel RJ-45 Kryss
Sender a1 - rot oder ohne Ring - 4 - 3
Sender b1 - schwarz oder 1 Ring - 5 - 6
Empfaenger a2 - weiss oder zwei Ringe m. grossem Abstand - 3 - 4
Empfaenger b2 - gelb oder zwei Ringe m. kleinem Abstand - 6 - 5
/ / /
/ / /
3 RX+ 2a --[100 Ohm]----+ ---------- / / ----------
4 TX+ 1a --[100 Ohm]--+ | | 87654321 | / | 12345678 |
5 TX- 1b -------------+ | |__ __|/ |/_ /_|
6 RX- 2b ---------------+ |____| |/___|
RJ-45 Stecker RJ-45 Buchse.
S.0-Anschlussklemmen
RX+ RX- TX+ TX-
3 6 4 5 fra NT-boks
2a 2b 1a 1b
| | | |
| +---(--- +40V- ---(----+ |
| | | | | |
(4) | (5) (3) | (6) fra ISDN-kort som
========= =========== NT-boks
========= ===========
| | | |
Send Motta
NT-Schaltung
/ / \ \
/ / \ \
/+-----+ +-----+\
+-+ 8 | | 1 +-+
| 7 | | 2 |
RX(-) | 6-----------------\ /-------------------3 | RX+
TX(-) | 5---------------\ \ / /-----------------4 | TX+
TX(+) | 4----------------------/ / /--------------5 | TX-
RX(+) | 3------------------------/ / /------------6 | RX-
| 2 | \ \-----/ / | 7 |
+-+ 1 | / \---------/ \ | 8 +-+
+-----+/ \+-----+
NT TA
RJ-45 Stecker RJ-45 Stecker
fra ISDN-kort til Telefon
Kryssing av kabel
Monteringen blir for ISDN:
RJ-45
Kabel par gren farge pinn. Signal/bruk gammel kobling
NT TA plint RJ-45
1 a blå-kvit 5 TX- RX- 5 3---+
1 b blå 4 TX+ RX+ 2 4-+ |
2 a orange-kvit 1 3 1 | |
2 b orange 2 4 2 | |
3 a grønn-kvit 3 RX+ TX+ 1 5-+ |
3 b grønn 6 RX- TX- 6 6---+
4 a brun-kvit 7 ADSL 7 7
4 b brun 8 ADSL 8 8
Monteringen blir for ETHERNET:
T568A RJ-45 T568B RJ-45 (X-over)
Kabel par gren farge pinn. Signal/bruk pinn Signal/bruk
PC HUB/SW PC
1 a blå-kvit 5 5
1 b blå 4 4
2 a orange-kvit 3 RX+ TX+ 1 TX+
2 b orange 6 RX- TX- 2 TX-
3 a grønn-kvit 1 TX+ RX+ 3 RX+
3 b grønn 2 TX- RX- 6 RX-
4 a brun-kvit 7 7
4 b brun 8 8
ELKO EP009 AT&T 568B AT&T 568A
plint RJ-45 pinn res ISDN ETHER CAT5 CAT5
1 5 | TX- 2a 1a
2 4 | TX+ 2b 1b
3 1 TX+ 3a 3a
4 2 TX- 1b 3b
5 3 | RX+ RX+ 1a 2a
6 6 | RX- RX- 3b 2b
7 7 4a 4a
8 8 4b 4b
AT&T 568- M11 B/C
plint RJ-45 pinn AT&T 568A AT&T 568B ISDN ETHER
1 1 3a 2a TX+
2 2 3b 2b TX-
3 3 2a 3a RX+
4 4 1b 1b
5 5 1a 1a
6 6 2b 3b RX-
7 7 4a 4a
8 8 4b 4b
ELKO EP013 (Tele)
plint RJ-45 pinn res ISDN
1 5 | TX-
2 4 1 TX+
3 3 RX+
4 6 o RX-
5 1 o
6 2
7 7
8 8
FA 670-C5E
plint RJ-45 pinn res ISDN ETHER T568A T568B
1 2 TX- 3b 2b
2 1 TX+ 3a 2a
3 4 TX+ 1b 1b
4 5 TX- 1a 1a
5 NC
6 NC
7 7 4a 4a
8 8 4b 4b
9 3 RX+ RX+ 2a 3b
10 6 RX- RX- 2b 3a
+-------------+
| 6 5 |
| 7-(7) 4-(5)|
| 8-(8) 3-(4)|
| 9-(3) 2-(1)|
|10-(6) 1-(2)|
|-------------|
| 87654321 | RJ-45
+-------------+
1.1 What is PBX4Linux?
PBX4Linux is pure software based ISDN PBX, that connects external lines, internal telephones, and optionally
voice over IP. It is designed to run with Linux only because it uses the kernel’s mISDN passive driver by
Karsten Keil. It can work together with OpenH323, that is a voice over IP implementation complied to the
H.323 ITU-T standard.
Only internal cards that plug into an ISA or PCI slot are supported. ISA Plug&Play cards are also supported, but need some
additional manual configuration by means of the isapnptools. For details on the configuration see question config_pnp.
Internal cards may be active, semi-active, or passive. Unless you have paid big money, assume you have a passive card.
More about the difference: see question hardware_activepassive.
Right now there is a driver for all passive card with certain Siemens chipsets (HiSax driver). Have a look at the
README.HiSax that comes with the driver for the most up to date information on supported cards and which parameter to pass
to Hisax. Here the status from 1st February 2002 (constantly improving):
Only internal cards that plug into an ISA or PCI slot are supported. ISA Plug&Play cards are also supported,
but need some additional manual configuration by means of the isapnptools. For details on the
configuration see question config_pnp.
Internal cards may be active, semi-active, or passive. Unless you have paid big money, assume you have a
passive card. More about the difference: see question hardware_activepassive.
Right now there is a driver for all passive card with certain Siemens chipsets (HiSax driver). Have a look at the
README.HiSax that comes with the driver for the most up to date information on supported cards and which
parameter to pass to Hisax. Here the status from 1st February 2002 (constantly improving): [liste over kort]
NT mode
When multiple devices are connected to the ISDN connection, then all user device behave as slaves,
where the network terminator (NT) behaves as master and synchronizes the communication on the S0 bus.
The special behavior of the network terminator is called NT mode. User devices are normally not capable
of running in NT mode. As a result, user devices can not communicate with each other even when they
are connected via a crossed cable. Only some special ISDN cards (HFC chipset) are capable of running in
NT mode, and can directly communicate with other ISDN user devices via a crossed cable.
The compatibility issue isn't a matter of the interface card having its own
NT1 or needing an external one. The ability of an ISDN interface to
communicate with another ISDN device is further upstream than the NT1. One
of the two devices must have its micro code configured to assume the master
role, NT or Network Termination, rather than the subordinate, TE or Terminal
Equipment, role that limits most pc interface cards. The NT device must be
capable of handling bits in the S/T data stream that are earmarked for
different purposes depending upon travel from NT to TE or TE to NT.
If you are reading this FAQ online, you may consider downloading the whole thing, and reading it offline (much cheaper).
To download the latest version of this FAQ in TXT/HTML/SGML format, go to the homepage of this FAQ:
