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  of old, vintage, and outdated    (.---.)-._.-.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.          '-----' sjw

  
  CONNECT
       .-----------------------.   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 SIP VoIP provider.

  There are many options for ATAs:

  ┌────────────┬───────────┬──────────────────────┐
  │Grandstream │ HT801/802 │ One-Port, Two-Port   │
  │            │ HT812     │ Two-Port with Router │
  │            │           │ User PDF Admin PDF   │
  ├────────────┼───────────┼──────────────────────┤
  │Cisco       │ SPA112    │ Two-Port             │
  │            │ SPA122    │ Two-Port with Router │
  │            │           │ Quick PDF Admin PDF  │
  ├────────────┼───────────┼──────────────────────┤
  │Cisco       │ IAD2400   │ 8, 16, 24 ports      │
  │            │           │ Info HW PDF SW 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 AT initilization 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 MT5600 PDF AT PDF
  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.
  
  ╔══════╗    ╔══════════╗     ╔══════════╗
  ║ 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 Asterisk PBX 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.
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