Free gateway 2.0 software for linux

The RSS-IM Gateway allows any site with an RSS feed to make their feed available via the IM network of their choice. It is very easy to set up.
Just create a new IM nick on your favorite network (links here) - something like "MySiteIM" always works - fill out the blanks in the config.im file, and your headlines are viewable on that network automatically, including from most cell phones if you use AIM or MSN. It works with AIM, ICQ, YIM and Jabber.
This project is a derivative of the SDBA Revolution IM Application Server version 1.85 and includes all files you will need. You will need to have an uptodate version of Perl with XML::RSS, in addition to the other libraries required.
Using:
- Make sure Perl is set up correctly, with all libraries (including XML::RSS!).
- Open config.im and put in your chosen nick and password for the bot, edit the network (default is AIM), and put in the URL to your feed.
- Start sdba.pl from the command line.
- Message the nick you chose from any IM client.

Gateway is a JavaEE application developed by the Vermont Department of Taxes. It provides a web services framework for accepting Streamlined Sales Tax registrations and returns.

The project also includes a web interface for manually submitting transmissions. The goal is to build an extensible framework upon which future tax services can be built.

SMBGate provides a gateway and firewall with authentication managed by a Samba Primary Domain Controller. The firewall rules are set on a per-user basis.

Wap-IRC gateway allows you to chat on IRC with any WAP-enabled mobile phone. It works by running the server program on a Unix-like system that is accessible over the Internet. It doesnt require phones which have programs written in Java.
I am Tomas Vilda, student studying at University of Vilnius, Lithuania. I started interested in wap when I bought my first wap enabled mobile phone. I searched for the running irc-wap gateway. But with no success. Actually, I found some, but no such, that I make to connect to specified IRC server. Then I found wapirc gateway written by Alan Cox . I tried it on my FreeBSD server. During compilation I got error, because of bad Makefile. So I compiled it by correcting Makefile and ran successful. My mobile is Siemens C55. To start using this wapirc gateway I have to make some modifications. They are listed below.
Enhancements:
- cgi/main():refresh included into main-cgi, saves us .htaccess and external wmlscript
- cgi/write_wml(): fixed bug in linked nicks with ACTIONs and * in there.
- server/parseinput(): Added /me Command for * ACTION in channels
- cgi/write_card(): get refresh timer from config.h
- server/removeallitems(): doing a list cleanup when disconnecting, so we can rejoin after quit without part
- server/print_out(): screenbuffer with latest messages, scrolling top down
- cgi/write_card(): reload on a regular basis, when not typing something in. New formular works on more phones, informative header
- cgi/wml_link()/wml_header(): wml1.2 compatible links/input field for accesskey element
- cgi/write_wml(): new linewise printout, italic makeup, nicks are linked setting defaultvalues for talking on channels and privat.
- cgi/main(): Expire header
- Makefile: make install

astfax allows you to create an email to fax gateway. It processes incoming email messages with attached fax images and sends them out via the txfax module. Asterisks outgoing call queue is used to initiate the fax call and transfer. The call file generated for the outgoing call queue is in template form, and can easily be updated for any systems configuration. It can be used (for example) with qmail, asterisk, spandsp, and eps to create an email/fax gateway.

For complete functioning of astfax you will need a mail solution which allows you to send an email via stdin to an external delivery program. Generally, qmail is the preferred MTA on Unix-based systems. One can easily set up an Asterisk-dedicated qmail MTA for delivery of voicemail
messages, faxes, etc.

This is the IRC gateway for Microsoft(tm) Messenger.

You probably wonder what this is and why I wrote this. Some more information can be found in the DESIGN document that is included in this package.

This software is written by Johannes Verelst and is licensed through the GPL license (http://www.gnu.org/licenses/gpl.html).

Note that this is pre-alpha code, it probably does a bit that you want, but dont email me bugreports or feature requests. It is under heavy development.

If this program doesnt run try to install corectly all required perl modules yourself.

The idea for this program came when I was getting annoyed by the fact that I had to use different programs for both IRC and MSN. Since I dont know of any good MSN clients that can do IRC too (on unix, ofcourse), I started thinking about the other way around: an IRC client that can do MSN.

When I saw a piece of software written by Adam Swann that allows easy connectivity to MSN with perl, I decided to write an IRC server in perl that would connect to MSN for you. This is the result.

Ekiga is a SIP and H.323 application, supporting audio and video, and is the successor of GnomeMeeting.
Main features:
- Call Forwarding on busy, no answer, always (SIP and H.323)
- Call Transfer (SIP and H.323)
- Call Hold (SIP and H.323)
- DTMFs support (SIP and H.323)
- Basic Instant Messaging (SIP)
- Text Chat (SIP and H.323)
- Possibility to register to several registrars (SIP) and gatekeepers (H.323)
- Possibility to use an outbound proxy (SIP) or a gateway (H.323)
- Message Waiting Indications (SIP)
- Audio AND Video (SIP and H.323)
- STUN support (SIP and H.323)
- DTMFs support
- LDAP support
Enhancements:
- MJPEG support (V4L1) was added.
- Workarounds were added for broken qspca driver (V4L1).
- The account strings check procedure was fixed.
- Auto build scripts were added for Win32.
- A Win32 installer was added.
- Unregistration from SIP registrars running on non standard ports was fixed.
- A possible problem with H.323 calls was fixed.
- Possible Avahi crashes were fixed.
- A possible routing problem for incoming calls initiated through uncompliant SIP INVITE PDUs was fixed.
- A crash on Win32 when exchanging SIP text messages was fixed.

YateClient is a next-generation telephony engine.
While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended.
Voice, video, data, and instant messaging can all be unified under Yates flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
Yate can be used as a:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- H.323SIP Proxy
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and/or client
- IP Telephony server and/or client
- Call center server
- IVR engine
- Prepaid and/or postpaid cards system