Information

Tips & Tricks on how to find a phone line, how to use your modem, and information about how everything works

Tips & Tricks on how to find a phone line, how to use your modem, and information about how everything works.

2600.network is a public service for dial-up users. Its purpose is to allow users of old, vintage, and outdated hardware to dial in with real modems to real systems.

CONNECT

Dial in with any modem connected to any kind of analog line you 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: **********

Alternatively, if you register via the web, there is an option to generate PPP credentials for TCP/IP connectivity.
There are 48 ports of dial-up access available.

Protocol	Speed
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	29333 - 56000 (49333 reliably)

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. For VoIP connections, use the G.711/ulaw codec along the entire path.

(International support is not disabled, but your mileage 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 SIP VoIP provider.

Users have had success with VoIP.ms as their carrier.

ATA Options
Grandstream	HT801/802 (One-Port, Two-Port), HT812 (Two-Port with Router)
Cisco	SPA112 (Two-Port), SPA122 (Two-Port with Router)
Cisco	IAD2400 (8, 16, 24 ports)
Patton	SmartNode (Many options)

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 AT initialization strings to manually set your modem speed. Connections have been made up to 52k in testing, your results may vary. Slower modem speeds might work better - but not neccesarily.

http://www.modemhelp.org/ has a database you can try. These modems are known to work:

MultiTech MT5600
USRobotics Courier V.Everything
Creative 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.

┌──────────┐  ┌──────────┐   ┌────────────┐ v TCP/IP                  
│ Computer │  │ Computer │   │ Patton RAS ├───────────┬─┬───────┬───┐ 
└─────────┬┘  └┬─────────┘   └┬───────────┘      SNMP>│ │       │ 8 │ 
    RS232>│    │<RS232        │<Digital (T1/PRI)    10│ │9     ┌┴┐ ┌┴┐
 ┌────────┴┐  ┌┴────────┐    ┌┴──────────┐  ┌─────────┴┐│      │P│ │T│
1│  Modem  │  │  Modem  │   6│ Cisco IAD │  │ InfluxDB ││      │P│ │e│
 └────────┬┘  └┬────────┘    └┬──────────┘  └─┬────────┘│      │P│ │l│
2  Analog>│    │<Analog       │<Digital(SIP)  │RADIUS>┌─┘      └┬┘ │n│
   ┌──────┴┐  ┌┴──────┐      ┌┴─────────┐HTTP>│ ┌─────┴┐  9a    │  │e│
   │ Telco │  │  ATA  │     5│ Asterisk ├─────│─┤ Auth ├────┐   │  │t│
   └──────┬┘  └┬──────┘      └┬─────────┘  11 │ └┬─────┘    │   │  └┬┘
          │    │<Digital(SIP) │<Digital(SIP)  │9b│<SQL HTTP>│  ┌┴───┴┐
          │   ┌┴──────────┐  ┌┴─────────┐  ┌──┴──┴─────┐    │  │ VPN │
 *Digital>│   │ Your VoIP │  │ My VoIP  │  │ Webserver │    │  └┬───┬┘
          │   │ Provider  │  │ Provider │  └┬──────────┘    │   │8 a│ 
          │   └┬──────────┘  └┬─────────┘ 12│<2600.network  │   │   │ 
3         │    │<Digital*    4│<Digital*    │TelnetBBSGuide>│   │   │ 
   ┌──────┴────┴──────────────┴─┐       ┌───┴───────────────┴───┴───┴┐
   │            PSTN            │       │          Internet          │
   └────────────────────────────┘       └────────────────────────────┘

The system is created by using a combination of technologies and hardware.

1.) First, your commands are sent to a modem over a serial line (Or maybe, you've got a modem built in to your device!)

2.) The modem uses an analog telephone line for it's phone service.
This is the only part in the chain which there is an Analog-to-Digital conversion.

3.) Whether you have a landline or an Analog Telephone Adapter using a VoIP connection, your call continues to the PSTN.
*This is most likely a digital connection.

4.) Then, it travels from the PSTN to my VoIP Provider.
*Also, most likely digital. If it weren't, this wouldn't work!

5.) The call moves from my VoIP provider via SIP to my server running Asterisk - a software driven Phone Branch Exchange.
This portion is digital - using the G.711/ulaw codec.

6.) The call is transferred from the Asterisk machine via SIP to a Cisco IAD-2432 24FXS, which converts the call to T1 - 24 phone lines for each of two ports.
This part has two different digital connections: SIP is G.711/ulaw, and T1/PRI is just the same type of Pulse Code Modulation over a different type of protocol.

7.) The call has now reached the Patton Remote Access Server at the core of the system, providing 64 DSPs, each with 2 modems. (although, it's only being supplied with two T1s - a maximum of 48 calls).

8.) The Patton connects these modems via telnet or rlogin to an IP/port combo, or via PPP to the internet.
8a.) The BBS and PPP connections are routed out a VPN from Private Internet Access.

9.) Authentication to the Patton is handled by a RADIUS database.
9a.) BBS logins are pulled automatically from TelnetBBSGuide.com.
9b.) PPP logins are pulled from the 2600.network web-registration system.
10.) Statistics are pulled via Telegraf SNMP agent into a InfluxDB database.
11.) Registration via Phone is handled between Asterisk and the Auth server.
12.) Statistics and logins are made avaible to users on the internet via the 2600.network website.