Dial-up pool

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Setting up your own Dial-up ISP with the ability to serve multiple clients is fairly straight forward but requires certain hardware, favorable network conditions, and a bit of patience.

This guide is intended to be as complete as possible and uses a Raspberry Pi computer for ease of use, but any computer with enough serial or USB ports that can run a modern Linux distribution will work.

TODO SWAP OUT IMAGE

DIY dial up ISP.jpg

Prerequisites

Hardware Requirements

  • One or more hardware modems (no software modems or "winmodems")
  • A Linux device (e.g. x86 computer, Raspberry Pi, etc.) to communicate with the modem(s) as the dial-in server
  • A client device with a modem (any type, hardware or software)
  • Some form of telephony connection between clients' modems and the ISP modems. I'll be using a software PBX and VoIP analog telephone adapters (ATAs)

The hardware used in this guide:

  • Raspberry Pi 3B
  • Modems:
    • Matrix MX 14.4k Modem using XECOM XE1414 module (V.32) (Datasheet)
    • USRobotics Sportster 0459 56k External Modem (only 28.8 up) (Manual)
    • Practical Peripherals PM144MT II Modem (V.32)
    • USRobotics USR5637 USB 56k Modem (V.92) (Datasheet)
  • 3x USB to RS-232 serial adapters, ch341 chipset
  • 3x DE-9 to DB-25 serial adapters
  • 4x Linksys SPA-2102 analog telephone adapter (ATA)

Software Requirements

  • Raspbian, Debian, Ubuntu, or any modern Debian based Linux distribution (other flavors will work but the configuration will be different)
  • ppp
  • getty
  • Asterisk

Choosing Modem Hardware

Modem hardware varies greatly, but this project doesn't require anything beyond standard protocols between your ISP and client modems (V.32, V.34, etc.).
This guide should work with any dial-up modem that presents itself as a serial device to the operating system, including cheap USB modems, ISA modems, PCI modems and of course external RS-232 serial modems.

Note: You will have a lot of trouble using a softmodem/winmodem! You are much better off using a hardware-based modem.

I'll be using three external serial modems + USB-to-RS-232 adapters and one USB modem. Using dedicated serial hardware has the advantage of being easy to troubleshoot and scaling up to dozens of lines (if you have enough desk space and USB ports).

Setting up the Dial-in Server

The dial-in server will answer calls from the modems and act as a proxy for access to network resources.

Using a Raspberry Pi

  1. Download the latest version of Raspbian Lite from raspberrypi.org and follow the installation instructions
  2. Before installing the SD card in your Raspberry Pi, enable SSH to avoid needed a mouse and keyboard:
    1. Mount the Raspberry Pi boot partition (most operating systems will do this automatically after writing the image)
    2. Create an empty file named ssh (with no extension) in the same folder as config.txt
    3. Safely eject the SD card
  3. Insert the SD card in your Raspberry Pi and connect the power and network cables
  4. Connect to the Raspberry Pi using SSH with the default username pi and password raspberry
    • If you know the IP address of the Pi, you can connect to it directly using your operating system's built-in SSH client (or PuTTY on Windows)
    • If you don't know the IP address of the Pi, use a utility like Adafruit's Pi Finder to find its IP address and log in

Using a PC or other hardware

Install a Debian-based Linux distribution of your choice before proceeding. Installation instructions are beyond the scope of this guide; you should consult your distribution's instructions for help (Ubuntu, Debian).

The Telephone Network

We need a way to connect our ISP modem to clients. There are many ways to approach this:

  1. Use the actual PSTN (i.e. real phone lines)
  2. Use a PBX to provide local connectivity
  3. Build your own circuity (not covered here as it would require extra configuration)
  4. Build a fake PSTN using VoIP ATAs and a software PBX

I've gone with the fourth option. Here's the breakdown:

  1. Asterisk - a VoIP PBX - is configured on the dial-in server to accept connections from multiple SIP client accounts and route calls between them.
  2. Cisco-Linksys SPA-2102 ATA - which supports two phone lines - is set up as both of those SIP clients connected to the PBX.
  3. The ISP-side modem(s) connect to one phone line, and a client device to a second line.

This design can scale up to as many modems and clients as desired, just add more ATAs!

Asterisk Setup

  1. Install asterisk
  2. sudo apt-get install asterisk
  3. Append configuration for the SIP clients to the end of /etc/asterisk/sip.conf
    [ata-modem1]
    context=default                 ; Using the default context because this is a simple design
    type=friend                     ; Allow calls to be placed and received to keep things simple
    secret=password                 ; Only the most secure passwords around here
    qualify=200                     ; Qualify peer is no more than 200ms away
    host=dynamic                    ; This device registers with us
    directmedia=yes                 ; Send RTP directly to the peer to reduce latency and jitter
    nat=no                          ; Only use symmetric IP routing
    
    [ata-modem2]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
    [ata-modem3]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
    [ata-modem4]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
    [ata-client1]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
    [ata-client2]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
    [ata-client3]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
    [ata-client4]
    context=default
    type=friend
    secret=password
    qualify=200
    host=dynamic
    directmedia=yes
    nat=no
    
  4. Edit /etc/asterisk/extensions.conf and make two changes:
    Search for [default] (should be around line 672) and comment out
    include => demo
    Underneath that line, add the new lines for the specific modems and the dial pool
    exten => 881,1,Dial(SIP/ata-modem1, 30)
    exten => 882,1,Dial(SIP/ata-modem2, 30)
    exten => 883,1,Dial(SIP/ata-modem3, 30)
    exten => 884,1,Dial(SIP/ata-modem4, 30)
    exten => _X!,1,Dial(SIP/ata-modem1&SIP/ata-modem2&SIP/ata-modem3&SIP/ata-modem4, 30)
    

    The _X! tells this dial plan rule to match any number a client dials and send the call to all of the ata-modem[1-4] clients simultaneously with a 30 second timeout. If you want to use a specific modem dial its extension and only that modem will ring.

  5. Enable the asterisk service so it starts on boot
    sudo systemctl enable asterisk
  6. Start Asterisk
    sudo systemctl start asterisk
  7. Open the Asterisk console to confirm your ATA lines are registered
    sudo asterisk -rvvvv
    raspberrypi*CLI> sip show peers
    Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
    ata-client1/ata-client1   10.1.0.126                               D  No         No             5060     OK (7 ms)                                    
    ata-client2/ata-client2   10.1.0.126                               D  No         No             5061     OK (7 ms)                                    
    ata-client3/ata-client3   10.1.0.125                               D  No         No             5060     OK (9 ms)                                    
    ata-client4/ata-client4   10.1.0.125                               D  No         No             5061     OK (10 ms)                                   
    ata-modem1/ata-modem1     10.1.0.108                               D  No         No             5060     OK (8 ms)                                    
    ata-modem2/ata-modem2     10.1.0.108                               D  No         No             5061     OK (7 ms)                                    
    ata-modem3/ata-modem3     10.1.0.128                               D  No         No             5060     OK (7 ms)                                    
    ata-modem4/ata-modem4     10.1.0.128                               D  No         No             5061     OK (8 ms)                                    
    8 sip peers [Monitored: 8 online, 0 offline Unmonitored: 0 online, 0 offline]
    

    If you make changes to your configuration after starting Asterisk, you can use the reload command in the console to reload the configuration.

ATA SIP registration example:

raspberrypi*CLI> 
[Apr 27 04:08:29] NOTICE[573]: chan_sip.c:24884 handle_response_peerpoke: Peer 'ata-client1' is now Reachable. (13ms / 200ms)
[Apr 27 04:08:29] NOTICE[573]: chan_sip.c:24884 handle_response_peerpoke: Peer 'ata-client2' is now Reachable. (5ms / 200ms)


Successful call to 888 (pool) example:

raspberrypi*CLI> 
  == Using SIP RTP CoS mark 5
       > 0x73a18d80 -- Strict RTP learning after remote address set to: 10.1.0.125:16386
    -- Executing [888@default:1] Dial("SIP/ata-client3-00000005", "SIP/ata-modem1&SIP/ata-modem2&SIP/ata-modem3&SIP/ata-modem4, 30") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Called SIP/ata-modem1
    -- Called SIP/ata-modem2
    -- Called SIP/ata-modem3
    -- Called SIP/ata-modem4
    -- SIP/ata-modem3-00000008 is ringing
    -- SIP/ata-modem1-00000006 is ringing
    -- SIP/ata-modem4-00000009 is ringing
    -- SIP/ata-modem2-00000007 is ringing
       > 0x73a31a78 -- Strict RTP learning after remote address set to: 10.1.0.108:16386
    -- SIP/ata-modem2-00000007 answered SIP/ata-client3-00000005
    -- Channel SIP/ata-modem2-00000007 joined 'simple_bridge' basic-bridge <3d2328ec-ef87-431e-be12-2dd9b84b6319>
    -- Channel SIP/ata-client3-00000005 joined 'simple_bridge' basic-bridge <3d2328ec-ef87-431e-be12-2dd9b84b6319>
       > Bridge 3d2328ec-ef87-431e-be12-2dd9b84b6319: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/ata-client3-00000005' and 'SIP/ata-modem2-00000007' - media will flow directly between them
       > 0x73a31a78 -- Strict RTP learning after remote address set to: 10.1.0.108:16386
       > 0x73a18d80 -- Strict RTP learning after remote address set to: 10.1.0.125:16386
[call proceeds then ends below]
    -- Channel SIP/ata-modem2-00000007 left 'native_rtp' basic-bridge <3d2328ec-ef87-431e-be12-2dd9b84b6319>
    -- Channel SIP/ata-client3-00000005 left 'native_rtp' basic-bridge <3d2328ec-ef87-431e-be12-2dd9b84b6319>
  == Spawn extension (default, 888, 1) exited non-zero on 'SIP/ata-client3-00000005'
       > 0x73a18d80 -- Strict RTP learning after remote address set to: 10.1.0.125:16386


ATA Configuration

Using a SPA-2102 on firmware 5.2.5 or 5.1.13 (Not carrier locked)

Example of configuring a client line. (Same settings for modem line, just a different username/password)

Lets start with factory default configuration and enabling web-based management.

  1. Connect a phone to LINE 1
  2. Dial **** to enter the configuration menu
  3. Dial 73738# then 1# then hang up. The unit is now factory reset
  4. Connect the Internet ethernet connection to your local network
  5. Dial **** to enter the configuration menu
  6. Dial 7932# then 1# then 1 then hang up. The web interface is now accessible from the 'Internet' side of the ATA
  7. Dial **** to enter the configuration menu
  8. Dial 110# to hear the IP address of your ATA

On a PC point your web browser at the IP of the ATA to load the web configuration. Several options need to be changed.

  1. Click "Admin Login"
  2. Click "Advanced"
  3. (Optiona) Click "WAN Setup" if you need to change the IP address of the ATA
  4. Click the "Voice" tab
  5. Click "Line 1"
  6. Change "Network Jitter Level" to "low"
  7. Change "Jitter Buffer Adjustment" to "disable"
  8. Set "Proxy" to the IP address of the Raspberry Pi running askterisk
  9. Set "User ID" to the username of the SIP user you are configuring, "ata-client3"
  10. Set "Password" to the password of that user
  11. Change "Call Waiting Serv" to "no"
  12. Change "Three Way Call Serv" to "no"
  13. Change "Preferred Codec" to "G711u"
  14. Change "Use Pref Codec Only" to "yes"
  15. Change "Silence Supp Enable" to "no"
  16. Change "Silence Threshold" to "high"
  17. Change "Echo Canc Enable" to "no"
  18. Change "Echo Canc Adapt Enable" to "no"
  19. Change "Echo Supp Enable" to "no"
  20. Change "FAX CED Detect Enable" to "no"
  21. Change "FAX CNG Detect Enable" to "no"
  22. Change "FAX Process NSE" to "no"
  23. Change "FAX Enable T38" to "no"
  24. Click "Submit All Changes"

Repeat for Line 2 if you need another line.

I won't go into much detail on ATA configuration since the topic has been beaten to death on various forums. This is the process to get basic communication:

  1. Set up both lines on the ATA to register to the PBX with usernames 'pap2t-ispmodem' and 'pap2t-client' and the password 'password'
  2. Use the G.711 µ-law codec
  3. Disable every echo cancellation option in your ATA (see here for PAP2T instructions)
  4. Set the jitter buffer to be as small as possible


Congratulations! You now have your own voice network.

The Dial-in Server

  1. Install your Debian-based Linux distribution of choice (not covered here)
  2. Update to latest packages and reboot if required
  3. sudo apt-get update
    sudo apt-get upgrade
    sudo reboot
  4. Connect the USB to RS-232 adapter and confirm it shows up as /dev/ttyUSBXXX (ls /dev/ to check). In my case, it presents as /dev/ttyUSB0
    My serial adapter is a "ID 1a86:7523 QinHeng Electronics HL-340 USB-Serial adaptor"
  5. Install ppp (and getty if your distro doesn’t have it by default)
    sudo apt-get install ppp mgetty
  6. Many of the old guides were written when inittab was still around, but it's 2019 and systemd has taken over.
    We need to create a systemd service for mgetty, so edit /lib/systemd/system/[email protected] (note the @) with your text editor of choice as root.
    [Unit]
    Description=External Modem %I
    Documentation=man:mgetty(8)
    Requires=systemd-udev-settle.service
    After=systemd-udev-settle.service
    
    [Service]
    Type=simple
    ExecStart=/sbin/mgetty /dev/%i
    Restart=always
    PIDFile=/var/run/mgetty.pid.%i
    
    [Install]
    WantedBy=multi-user.target
    
  7. Configure mgetty by editing /etc/mgetty/mgetty.config with your text editor of choice as root.
    Comment out everything except the debug level, and append the section for configuring the serial device:
    debug 9
    
    port ttyUSB0
     port-owner root
     port-group dialout
     port-mode 0660
     data-only yes
     ignore-carrier no
     toggle-dtr yes
     toggle-dtr-waittime 500
     rings 1
     speed 115200
     modem-check-time 160
    
    port ttyUSB1
     port-owner root
     port-group dialout
     port-mode 0660
     data-only yes
     ignore-carrier no
     toggle-dtr yes
     toggle-dtr-waittime 500
     rings 2
     speed 115200
     modem-check-time 60
    
    port ttyUSB2
     port-owner root
     port-group dialout
     port-mode 0660
     data-only yes
     ignore-carrier no
     toggle-dtr yes
     toggle-dtr-waittime 500
     rings 3
     speed 115200
     modem-check-time 60
    
    port ttyACM0
     port-owner root
     port-group dialout
     port-mode 0660
     data-only yes
     ignore-carrier no
     toggle-dtr yes
     toggle-dtr-waittime 500
     rings 4
     speed 115200
     modem-check-time 60
    
    
  8. Enable the mgetty service so it starts on boot:
    sudo systemctl enable [email protected]
    sudo systemctl enable [email protected]
    sudo systemctl enable [email protected]
    sudo systemctl enable [email protected]

  9. Start mgetty:
    sudo systemctl start [email protected]
    sudo systemctl start [email protected]
    sudo systemctl start [email protected]
    sudo systemctl start [email protected]
  10. Configure ppp by editing /etc/ppp/options
    Like above, comment out everything except these settings:
    # Define the DNS server for the client to use
    ms-dns 8.8.8.8
    # async character map should be 0
    asyncmap 0
    # Require authentication
    auth
    # Use hardware flow control
    crtscts
    # We want exclusive access to the modem device
    lock
    # Show pap passwords in log files to help with debugging
    show-password
    # Require the client to authenticate with pap
    +pap
    # If you are having trouble with auth enable debugging
    debug
    # Heartbeat for control messages, used to determine if the client connection has dropped
    lcp-echo-interval 30
    lcp-echo-failure 4
    # Cache the client mac address in the arp system table
    proxyarp
    # Disable the IPXCP and IPX protocols.
    noipx
    
  11. Create a device option file by editing /etc/ppp/options.ttyUSB0
    local
    lock
    nocrtscts
    192.168.32.1:192.168.32.2
    netmask 255.255.255.252
    noauth
    proxyarp
    lcp-echo-failure 60
    


    /etc/ppp/options.ttyUSB1

    local
    lock
    nocrtscts
    192.168.32.5:192.168.32.6
    netmask 255.255.255.252
    noauth
    proxyarp
    lcp-echo-failure 60
    

    /etc/ppp/options.ttyUSB2

    local
    lock
    nocrtscts
    192.168.32.9:192.168.32.10
    netmask 255.255.255.252
    noauth
    proxyarp
    lcp-echo-failure 60
    

    /etc/ppp/options.ACM0

    local
    lock
    nocrtscts
    192.168.32.13:192.168.32.14
    netmask 255.255.255.252
    noauth
    proxyarp
    lcp-echo-failure 60
    
  12. Create the user for PAP authentication:
    sudo useradd -G dialout,dip,users -m -g users -s /usr/sbin/pppd dial
  13. Set a password:
    sudo passwd dial
  14. (I used dial, same as the username)
  15. Edit /etc/ppp/pap-secrets and append the username and password (same as you entered above, quotes included):
    dial * "dial" *
  16. Enable packet forwarding for IP4 by editing /etc/sysctl.conf:
    net.ipv4.ip_forward=1
  17. The last step for the dial-up server is to configure the firewall to allow traffic forwarding from PPP out onto the network (and off to the Internet).
    1. On Linux distributions with iptables, you need to add a line to /etc/rc.local to enable masquerading. If your Ethernet interface is named eth0, you would add this line:
      iptables -t nat -A POSTROUTING -s 192.168.32.0/24 -o eth0 -j MASQUERADE
    2. On modern Ubuntu installs, ufw is used as a frontend to iptables, so the procedure is a bit different. Follow this guide, but you can omit -o eth0 and use -s 192.168.32.0/24.

Troubleshooting

When using an external modem, the choice of USB to RS-232 adapter seems to be crucial. There aren't many requirements, but you must use an adapter that supports hardware flow control. If you need to purchase an adapter, you can either get one that explicitly says it supports hardware flow control ($$$), or play the eBay lottery and buy a half-dozen different models and hope one of them works.

I ran into a bug in Debian 9.5 with my USB to serial adapter using the ch341 driver, where setting the baud rate was not working on some Linux kernels. (Seems to be this bug)

To troubleshoot modem communication and baud rate settings, use minicom (or screen) to open a session over serial and try different settings (or read your modem's manual!). Sending the command 'AT' followed by a new line should result in your modem replying 'OK'.
If you're getting nothing at all out of your modem, perform a serial loopback test

If mgetty is not answering incoming calls, it may be having trouble communicating with your modem. Check the logs in /var/log/mgetty/ to determine the problem. You may need to set a modem initialization string in the mgetty device config file, so check your modem's manual for help on this.