2600.network is a public service for dial-up users
It's purpose is to allow users _____
of old, vintage, and outdated (.---.)-._.-.
hardware to dial in with real /:::\ _.---'
modems to real systems. '-----' sjwCONNECT
.-----------------------. Dial in with any
/=======================/| modem connected
/=======================/ | to any kind of
/_______________________/ / analog line you
/ /_// // // // // usr/ / can find.
:-----------------------: /
|_______________________|/ VoIP is 100% OK!
Once you dial in, use a username/password combo
from the logins page, and it will redirect you
to a telnet BBS of your choice.
#################################################
############---https://2600.network---###########
#################################################
lOgiN:
uSerNaME: 20forbeers
PaSsWorD: **********
There are 48 ports of dial-up access available.
┌───────────────────────────────────────────────┐
│ Modems can connect with these protocols: │
├────────────┬──────────────────────────────────┤
│ v.21 │ 300 │
│ v.22(bis) │ 1200 - 2400(bis) │
│ v.23 │ 1200 │
│ v.32(bis) │ 4800 - 9600 14400(bis) │
│ v.34 │ 9600 - 33600 │
│ v.90 │ 29333 - 56000 49333 reliably │
│ k56flex │ 32000 - 56000 │
│ v.92 │ disabled │
└────────────┴──────────────────────────────────┘
CONNECT / TELEPHONE
You should be able to dial into 2600.network
from any PSTN connected line in the US/Canada.
This includes POTS landlines and VoIP
(International support is not disabled, but your
milegae may vary. Regional servers are being
considered, please contact us or donate to help.)
Don't want to pay for a landline? No worries! You
don't have to! Using an Analog Telephone Adapter,
you can dial in to 2600.network. The ATA will
give you analog service from a SIPVoIP provider.
There are many options for ATAs:
┌────────────┬───────────┬──────────────────────┐
│Grandstream │ HT801/802 │ One-Port, Two-Port │
│ │ HT812 │ Two-Port with Router │
│ │ │ User PDFAdmin PDF │
├────────────┼───────────┼──────────────────────┤
│Cisco │ SPA112 │ Two-Port │
│ │ SPA122 │ Two-Port with Router │
│ │ │ Quick PDFAdmin PDF │
├────────────┼───────────┼──────────────────────┤
│Cisco │ IAD2400 │ 8, 16, 24 ports │
│ │ │ InfoHW PDFSW PDF │
├────────────┼───────────┼──────────────────────┤
│Patton │ SmartNode │ Many options │
└────────────┴───────────┴──────────────────────┘
Use the G.711/ulaw codec along the entire path.
CONNECT / MODEMS
USB-to-Serial adapters work just fine, but you may
(or may not) need to disable FIFO buffers inside
of the COM port properties in Device Manager.
Make sure your serial port speed is set to the
maximum speed allowed by your modem! Otherwise,
your modem speed might be affected, or become
unable to negotiate. This includes inside of device
manager, and inside of the application you use.
Use ATinitilization strings to manually set your
modem speed. Connections have been made up to 49.3k
in testing, your results may vary. Slower modem
speeds will work better.
http://www.modemhelp.org/ has a database you can try,
or you can use some of our own AT strings, manuals,
and AT command sets gathered by 2600.network:
MultiTech MT5600PDFAT PDFUSRobotics Courier V.EverythingCreative ModemBlaster 56k (EU)CONNECT / EXAMPLES
Examples of users successfully connecting will be
put here. Or, go check out the stats page.
ADMIN
How is this possible?
The entire system can be understood by learning
how the G.711/ulaw codec works. It's the backbone
behind T1/DS1/ISDN PRI voice lines that have been
powering the Public Switched Telephone Network
for years.
This system is possible because the amount of
analog-to-digital and digital-to-analog conversion
is limited; allowing the G.711/ulaw codec to be
passed along the route unaltered.
╔══════╗ ╔══════════╗ ╔══════════╗
║ PSTN ║────║ VoIP ║─────║ Cloud ║
╚══════╝ ║ Proivder ║ SIP ║ Asterisk ║
╚══════════╝ ╚══════════╝
┌───────────IAX2─────────────┘
╔══════════╗ ╔═══════╗
║ Local ║ SIP ║ Cisco ║ T1 ╔════════╗
║ Asterisk ║─────║ IAD ║─────║ ║
╠══════════╣ T1 ╚═══════╝ ║ Patton ║
║ TE420 ║─────────┘ T1 ║ 2960 ║
║ DAHDI ║───────────────────║ ║
╚══════════╝ ╚════════╝
┌─────── TELNET,RLOGIN ──────┘ │ S│N
╔═════╗ ┌─── RADIUS, Syslog ───┘ M│P
║ BBS ║ ╔═══════════╗ ╔═══════╗
╚═════╝ ║ Auth/Logs ║ ║ WWW ║
╚═══════════╝ ╚═══════╝
The system is created by using a combination of
technologies and hardware.
First, your call comes in from the PSTN. Then, it
travels through a VoIP Proivder (SIP) to a machine
in the cloud running Asterisk, a software driven
Phone Branch Exchange.
The call then travels (IAX2) to a local server
running Asterisk. This server fowards your call
one of two ways.
One way (SIP) is to the Cisco IAD-2432 24FXS,
which converts (T1) the call sent to the Patton.
Another route is provided directly (T1) from the
local AsteriskPBX to the Patton. A PCIe card, the
Digium TE420, and it's drivers/software DAHDI give
this capability.
You have now reached the Patton 2960 Remote Access
Server at the core of the system, providing 24 DSPs,
each with 2 modems. The Patton connects these modems
via telnet or rlogin to an IP/port combo within it's
local database or remote RADIUS database.