From c6c05c6f47c8294222f87fb053e858cdd9c9ff12 Mon Sep 17 00:00:00 2001 From: Jason McCormick Date: Wed, 11 Dec 2024 09:53:43 -0500 Subject: [PATCH] Deployed 1004b90 with MkDocs version: 1.6.1 --- 404.html | 23 + adv-topics/allowdenylists/index.html | 23 + adv-topics/audio-level/index.html | 23 + adv-topics/autopatch/index.html | 23 + adv-topics/broadcastify/index.html | 23 + adv-topics/cloud-kernel/index.html | 23 + adv-topics/commands/index.html | 23 + adv-topics/conftmpl/index.html | 23 + adv-topics/hotspot_hd/index.html | 23 + adv-topics/httpreg/index.html | 23 + adv-topics/iaxreg/index.html | 23 + adv-topics/incompatibles/index.html | 23 + adv-topics/multinodesnetwork/index.html | 2408 ++++++++++++++++++++++ adv-topics/noderesolution/index.html | 23 + adv-topics/other-software/index.html | 23 + adv-topics/permissions/index.html | 23 + adv-topics/sa818modules/index.html | 23 + adv-topics/sip-phone/index.html | 23 + adv-topics/tts/index.html | 29 +- adv-topics/uefi-secureboot/index.html | 23 + adv-topics/usbinterfaces/index.html | 23 + allmon3/basics/index.html | 23 + allmon3/config/index.html | 23 + allmon3/install/index.html | 23 + allmon3/usermgmt/index.html | 23 + developers/aptly/index.html | 23 + developers/package-builds/index.html | 29 +- developers/source-install/index.html | 23 + index.html | 23 + mans/asl-find-sound/index.html | 23 + mans/asl-node-lookup/index.html | 23 + mans/asl-play-arn/index.html | 23 + mans/asl-say/index.html | 23 + mans/asl-setup-dkms-mok/index.html | 23 + pi/cockpit-basics/index.html | 23 + pi/cockpit-console/index.html | 23 + pi/cockpit-firewall/index.html | 23 + pi/cockpit-get-started/index.html | 23 + pi/cockpit-network/index.html | 23 + pi/cockpit-services/index.html | 23 + pi/cockpit-updates/index.html | 23 + print_page/index.html | 199 +- search/search_index.json | 2 +- sitemap.xml | 106 +- sitemap.xml.gz | Bin 630 -> 640 bytes user-guide/asl-cmds-tools/index.html | 23 + user-guide/betatest/index.html | 23 + user-guide/index.html | 23 + user-guide/install/index.html | 23 + user-guide/known-issues-diffs/index.html | 23 + user-guide/menu/index.html | 23 + user-guide/passwords/index.html | 23 + user-guide/pi-detailed/index.html | 23 + user-guide/support/index.html | 23 + user-guide/troubleshooting/index.html | 23 + user-guide/updates/index.html | 23 + user-guide/whats-new/index.html | 23 + 57 files changed, 3864 insertions(+), 59 deletions(-) create mode 100755 adv-topics/multinodesnetwork/index.html diff --git a/404.html b/404.html index bc08280..dbd6858 100755 --- a/404.html +++ b/404.html @@ -1323,6 +1323,8 @@ + + @@ -1758,6 +1760,27 @@ + + + + + + +
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    Multiple Nodes on the Same Network

    +

    Multiple AllStarLink nodes on the same IP subnet (i.e. your "local network") when +those nodes are behind a NAT device can be a challenge. The reason is that +AllStarLink works exclusively with public IP addresses. When two nodes +are on the same network behind a NAT/router they independently need to +use a public IP address with the AllStarLink system but need to know about each +other's local IPs rather than their shared public IP. Additionally both nodes +cannot use the same UDP port as they will fight over it with the NAT/router.

    +

    For this example, let's consider two nodes on the same subnet. The nodes are:

    + + + + + + + + + + + + + + + + + +
    NodeLocal IP
    630010192.168.0.10
    630011192.168.0.11
    +

    These nodes are behind a NAT/router with a public IP address of 203.0.113.186.

    +

    When each node connects and registers with AllStarLink, its perceived registration +IP address will be 203.0.113.186. If another node asks "where is node 630010" it +would get back the answer 203.0.113.186. However, this is also the case for the +internal nodes. If node 630011 asks "where is node 630010" it also gets back +203.0.113.186. For various network reasons, this will not work.

    +

    One also has to consider the UDP port. By default, each Asterisk system +uses the default UDP port of 4569. This is a published standard. However, through +a NAT device, only one system can use the port at a time. Thus our local +nodes 630010 and 630011 will fight over that port.

    +

    The make two (or more) nodes on the same network function, the following +steps must be taken:

    +
      +
    1. Assign each node's server a different IAX port
      2. +
    2. Tell each node's server about the other node's internal address and port
      3. +
    +

    In this example, node 630011 will be assigned a different UDP port. The table +then of the nodes is:

    + + + + + + + + + + + + + + + + + + + + +
    NodeLocal IPPort
    630010192.168.0.10UDP 4569
    630011192.168.0.11UDP 4570
    +

    This document assumes one knows how to forward the port internally for one's +router/NAT device and that is has already been done.

    +

    Assigning a Different IAX Port

    +

    There are two places where the IAX port must be changed when using a +non-standard port:

    +
      +
    1. /etc/asterisk/iax.conf
      2. +
    2. The AllStarLink Portal for that node's server
      3. +
    +

    Changing iax.conf

    +

    The best way to edit the IAX port is to use asl-menu. Run sudo asl-menu. +Choose 1 Node Settings and then 4 Update Asterisk IAX port. Set +the new UDP port in the box and hit OK. Then choose 2 Restart Asterisk. +Then close asl-menu.

    +

    Alternatively, edit the file directly. As root (i.e. sudo -s), edit the file /etc/asterisk/iax.conf. Find the line:

    +
    bindport = 4569
+
    +

    and change it to a different port. The ASL3 appliance comes preconfigured to permit +traffic on any port between 4560 and 4580. It is recommended to choose a port in +this range for any alternative IAX port. In this example, the node 630011 will be +changed to port 4570. Edit the file to state:

    +
    bindport = 4570
+
    +

    and save and close the file. Restart asterisk with systemctl restart asterisk.

    +

    Changing the Portal

    +
    +

    Server-Node Relationship

    +

    Following the operating model, the base Asterisk/app_rpt installation is the "Server" +and a "Server" hosts one or more "Nodes". Make sure that each node is assigned to a +different server if the nodes are configured on different installations.

    +
    +

    Do the following to change a node's server's IAX port.

    +
      +
    1. +

      Logon to https://www.allstarlink.org using your AllStarLink username +and password.

      +
      2. +
    2. +

      Click on Portal and then Node Settings.

      +
      3. +
    3. +

      In the table, find the Server for the node you want to edit. In this example +node 630011 is located on server node63011. Remember or jot down the server +name.

      +
      4. +
    4. +

      Click on Portal and then Server Settings.

      +
      5. +
    5. +

      Click on the server from step #4 (in this case "node630011")

      +
      6. +
    6. +

      In the field IAX Port change the port number to the port configured +in "Changing iax.conf" above. In this example, the port will be 4570.

      +
      7. +
    7. +

      Click Submit

      +
      8. +
    +

    Propagation of the change takes 1-2 minutes.

    +

    Telling Local Nodes About Their Neighbors

    +

    Each node on the same network needs to be hardcoded with information +about the other node. In this example, node 630010 needs to be hardcoded +for where 630011 is and vice-versa.

    +

    Configuring Node 630010

    +

    As root (i.e. sudo -s) edit the file /etc/asterisk/rpt.conf. Locate +the line with the node's own definition. For example:

    +
    630010 = radio@127.0.0.1/630010,NONE
+
    +

    After this line, add its local neighbor by its IP address and the alternative +IAX port. In this example, add 630011 as follows:

    +
    630010 = radio@127.0.0.1/630010,NONE
+630011 = radio@192.168.0.11:4570/630011,NONE
+
    +

    Note the inclusion of the port as :4570. This pattern can be repeated for multiple nodes +on the same network. Save the file and restart asterisk with systemctl restart asterisk.

    +

    Configuring Node 630011

    +

    The other node repeats same process but in the opposite direction. +As root (i.e. sudo -s) edit the file /etc/asterisk/rpt.conf. Locate +the line with the node's own definition. For example:

    +
    630011 = radio@127.0.0.1/630011,NONE
+
    +

    After this line, add its local neighbor by its IP address and the standard +IAX port since it is unchanged. In this example, add 630010 as follows:

    +
    630011 = radio@127.0.0.1/630011,NONE
+630010 = radio@192.168.0.10:4569/630010,NONE
+
    +

    Note the inclusion of the port as :4569. This pattern can be repeated for multiple nodes +on the same network. Save the file and restart asterisk with systemctl restart asterisk.

    +

    Testing Linking Locally

    +

    Test linking the two nodes together to ensure configuration. In this example, node +630010 will connect to node 630011 using the Asterisk CLI:

    +
    rpt cmd 630010 ilink 3 630011
+
    +

    Then the link should be shown to 630011 with the command:

    +
    rpt nodes 630010
+
    +

    that should display:

    +
    node630010*CLI> rpt nodes 630010
+
+************************* CONNECTED NODES *************************
+
+T630011
+
    +

    Disconnect the links with:

    +
    rpt cmd 630010 ilink 1 630011
+
    +

    After that, everything should work. The most common issue with this configuration +is incorrect router/NAT port forwarding or a typo in rpt.conf.

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    Reporting Bugs

    - +
    Next
    - Building ASL3-Asterisk + Multiple Nodes on the Same Network
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    Executing the GitHub Acti diff --git a/index.html b/index.html index f2dc2f6..3389607 100755 --- a/index.html +++ b/index.html @@ -1355,6 +1355,8 @@ + + @@ -1790,6 +1792,27 @@ + + + + + + +
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    • + + + + diff --git a/pi/cockpit-console/index.html b/pi/cockpit-console/index.html index db3e092..d66c6cc 100755 --- a/pi/cockpit-console/index.html +++ b/pi/cockpit-console/index.html @@ -1368,6 +1368,8 @@ + + @@ -1803,6 +1805,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + diff --git a/pi/cockpit-firewall/index.html b/pi/cockpit-firewall/index.html index 9c1cec0..3de9285 100755 --- a/pi/cockpit-firewall/index.html +++ b/pi/cockpit-firewall/index.html @@ -1368,6 +1368,8 @@ + + @@ -1803,6 +1805,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + diff --git a/pi/cockpit-get-started/index.html b/pi/cockpit-get-started/index.html index d27d35a..f4301f4 100755 --- a/pi/cockpit-get-started/index.html +++ b/pi/cockpit-get-started/index.html @@ -1407,6 +1407,8 @@ + + @@ -1842,6 +1844,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + diff --git a/pi/cockpit-network/index.html b/pi/cockpit-network/index.html index db63b6e..8bd6a2e 100755 --- a/pi/cockpit-network/index.html +++ b/pi/cockpit-network/index.html @@ -1368,6 +1368,8 @@ + + @@ -1803,6 +1805,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + diff --git a/pi/cockpit-services/index.html b/pi/cockpit-services/index.html index 24d4a33..67f601d 100755 --- a/pi/cockpit-services/index.html +++ b/pi/cockpit-services/index.html @@ -1368,6 +1368,8 @@ + + @@ -1803,6 +1805,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + diff --git a/pi/cockpit-updates/index.html b/pi/cockpit-updates/index.html index 7164eed..9ec70de 100755 --- a/pi/cockpit-updates/index.html +++ b/pi/cockpit-updates/index.html @@ -1368,6 +1368,8 @@ + + @@ -1803,6 +1805,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + diff --git a/print_page/index.html b/print_page/index.html index cfcc98a..b462ebc 100755 --- a/print_page/index.html +++ b/print_page/index.html @@ -1336,6 +1336,8 @@ + + @@ -1771,6 +1773,27 @@ + + + + + + +
  • + + + + + Multiple Nodes on the Same Network + + + + +
    • + + + + @@ -5436,7 +5459,181 @@

    Voices

    Note, since all files are squashed down to 8K uLaw format, there is no value in the "medium" or "high" quality models. Always use the low quality model.

    Reporting Bugs

    -

    Report bugs to https://github.com/AllStarLink/asl3-tts/issues

    Ended: Advanced Topics

    +

    Report bugs to https://github.com/AllStarLink/asl3-tts/issues

    Multiple Nodes on the Same Network

    +

    Multiple AllStarLink nodes on the same IP subnet (i.e. your "local network") when +those nodes are behind a NAT device can be a challenge. The reason is that +AllStarLink works exclusively with public IP addresses. When two nodes +are on the same network behind a NAT/router they independently need to +use a public IP address with the AllStarLink system but need to know about each +other's local IPs rather than their shared public IP. Additionally both nodes +cannot use the same UDP port as they will fight over it with the NAT/router.

    +

    For this example, let's consider two nodes on the same subnet. The nodes are:

    + + + + + + + + + + + + + + + + + +
    NodeLocal IP
    630010192.168.0.10
    630011192.168.0.11
    +

    These nodes are behind a NAT/router with a public IP address of 203.0.113.186.

    +

    When each node connects and registers with AllStarLink, its perceived registration +IP address will be 203.0.113.186. If another node asks "where is node 630010" it +would get back the answer 203.0.113.186. However, this is also the case for the +internal nodes. If node 630011 asks "where is node 630010" it also gets back +203.0.113.186. For various network reasons, this will not work.

    +

    One also has to consider the UDP port. By default, each Asterisk system +uses the default UDP port of 4569. This is a published standard. However, through +a NAT device, only one system can use the port at a time. Thus our local +nodes 630010 and 630011 will fight over that port.

    +

    The make two (or more) nodes on the same network function, the following +steps must be taken:

    +
      +
    1. Assign each node's server a different IAX port
      2. +
    2. Tell each node's server about the other node's internal address and port
      3. +
    +

    In this example, node 630011 will be assigned a different UDP port. The table +then of the nodes is:

    + + + + + + + + + + + + + + + + + + + + +
    NodeLocal IPPort
    630010192.168.0.10UDP 4569
    630011192.168.0.11UDP 4570
    +

    This document assumes one knows how to forward the port internally for one's +router/NAT device and that is has already been done.

    +

    Assigning a Different IAX Port

    +

    There are two places where the IAX port must be changed when using a +non-standard port:

    +
      +
    1. /etc/asterisk/iax.conf
      2. +
    2. The AllStarLink Portal for that node's server
      3. +
    +

    Changing iax.conf

    +

    The best way to edit the IAX port is to use asl-menu. Run sudo asl-menu. +Choose 1 Node Settings and then 4 Update Asterisk IAX port. Set +the new UDP port in the box and hit OK. Then choose 2 Restart Asterisk. +Then close asl-menu.

    +

    Alternatively, edit the file directly. As root (i.e. sudo -s), edit the file /etc/asterisk/iax.conf. Find the line:

    +
    bindport = 4569
+
    +

    and change it to a different port. The ASL3 appliance comes preconfigured to permit +traffic on any port between 4560 and 4580. It is recommended to choose a port in +this range for any alternative IAX port. In this example, the node 630011 will be +changed to port 4570. Edit the file to state:

    +
    bindport = 4570
+
    +

    and save and close the file. Restart asterisk with systemctl restart asterisk.

    +

    Changing the Portal

    +
    +

    Server-Node Relationship

    +

    Following the operating model, the base Asterisk/app_rpt installation is the "Server" +and a "Server" hosts one or more "Nodes". Make sure that each node is assigned to a +different server if the nodes are configured on different installations.

    +
    +

    Do the following to change a node's server's IAX port.

    +
      +
    1. +

      Logon to https://www.allstarlink.org using your AllStarLink username +and password.

      +
      2. +
    2. +

      Click on Portal and then Node Settings.

      +
      3. +
    3. +

      In the table, find the Server for the node you want to edit. In this example +node 630011 is located on server node63011. Remember or jot down the server +name.

      +
      4. +
    4. +

      Click on Portal and then Server Settings.

      +
      5. +
    5. +

      Click on the server from step #4 (in this case "node630011")

      +
      6. +
    6. +

      In the field IAX Port change the port number to the port configured +in "Changing iax.conf" above. In this example, the port will be 4570.

      +
      7. +
    7. +

      Click Submit

      +
      8. +
    +

    Propagation of the change takes 1-2 minutes.

    +

    Telling Local Nodes About Their Neighbors

    +

    Each node on the same network needs to be hardcoded with information +about the other node. In this example, node 630010 needs to be hardcoded +for where 630011 is and vice-versa.

    +

    Configuring Node 630010

    +

    As root (i.e. sudo -s) edit the file /etc/asterisk/rpt.conf. Locate +the line with the node's own definition. For example:

    +
    630010 = radio@127.0.0.1/630010,NONE
+
    +

    After this line, add its local neighbor by its IP address and the alternative +IAX port. In this example, add 630011 as follows:

    +
    630010 = radio@127.0.0.1/630010,NONE
+630011 = radio@192.168.0.11:4570/630011,NONE
+
    +

    Note the inclusion of the port as :4570. This pattern can be repeated for multiple nodes +on the same network. Save the file and restart asterisk with systemctl restart asterisk.

    +

    Configuring Node 630011

    +

    The other node repeats same process but in the opposite direction. +As root (i.e. sudo -s) edit the file /etc/asterisk/rpt.conf. Locate +the line with the node's own definition. For example:

    +
    630011 = radio@127.0.0.1/630011,NONE
+
    +

    After this line, add its local neighbor by its IP address and the standard +IAX port since it is unchanged. In this example, add 630010 as follows:

    +
    630011 = radio@127.0.0.1/630011,NONE
+630010 = radio@192.168.0.10:4569/630010,NONE
+
    +

    Note the inclusion of the port as :4569. This pattern can be repeated for multiple nodes +on the same network. Save the file and restart asterisk with systemctl restart asterisk.

    +

    Testing Linking Locally

    +

    Test linking the two nodes together to ensure configuration. In this example, node +630010 will connect to node 630011 using the Asterisk CLI:

    +
    rpt cmd 630010 ilink 3 630011
+
    +

    Then the link should be shown to 630011 with the command:

    +
    rpt nodes 630010
+
    +

    that should display:

    +
    node630010*CLI> rpt nodes 630010
+
+************************* CONNECTED NODES *************************
+
+T630011
+
    +

    Disconnect the links with:

    +
    rpt cmd 630010 ilink 1 630011
+
    +

    After that, everything should work. The most common issue with this configuration +is incorrect router/NAT port forwarding or a typo in rpt.conf.

    Ended: Advanced Topics

    Developer Guide ↵

    diff --git a/search/search_index.json b/search/search_index.json index caa450d..3c81488 100755 --- a/search/search_index.json +++ b/search/search_index.json @@ -1 +1 @@ -{"config":{"lang":["en"],"separator":"[\\s\\-]+","pipeline":["stopWordFilter"]},"docs":[{"location":"","title":"Welcome to AllStarLink 3","text":"

    AllStarLink\u2019s version 3 is the next generation of AllStar repeater and hotspot software. This version of AllStar has been redesigned to run on Asterisk 20 LTS (Long Term Support), the latest Debian Linux operating system, and modern hardware or virtual machines. The update from Asterisk 1.4 to Asterisk 20 implements over 15 years of Asterisk bug fixes, security improvements and enhancements. It brings with it the latest Asterisk applications, channel drivers and additional functionality.

    This update required app_rpt (the Asterisk application that is AllStar) and it's various modules to be heavily modified. Many memory leaks have been addressed, modules load or refresh more reliably and many bugs have been squashed. This all adds up to improved stability and uptime.

    In addition to the many app_rpt improvements and fixes, the code base is easier to maintain and enhance. The goal being to make app_rpt code accessible to more AllStar developers. Also, the app_rpt code base has been modified to meet Asterisk\u00ae coding guidelines.

    Supported Hardware & Software

    "},{"location":"#new-features","title":"New Features","text":"

    In addition to the app_rpt code update, we have added many new features to make ASL3 the best AllStar release yet.

    "},{"location":"adv-topics/allowdenylists/","title":"Allow and Deny Lists","text":"

    These changes can easily be made from the ASL3 menu. Here's how to change and view the lists at the Asterisk CLI should you choose to do so. These lists were previously referred to as whitelist and blacklist.

    ASL3 provides a way to establish either an \"allowlist\" or a \"denylist\" to limit access to your node(s). New to ASL3 is the capability for nodes on the same server to have different lists. Unlike ASL2, no modifications or scripts are required to the configuration files and there is no need to restart or reload the server. All changes are made to the Asterisk database. As soon as you make a change it takes effect.

    The \"allowlist\" allows inbound connects and blocks all others. Nodes on the same server are always allowed. If any nodes are on the \"allowlist\" the \"denylist\" is ignored.

    The \"denylist\" prevents nodes from connecting to your node. Add a \"denylist\" if you only have a few nodes you want to block.

    To see what's in the database.

    To delete an entire \"allowlist\"

    "},{"location":"adv-topics/audio-level/","title":"Calibrating Audio Levels For The Allstar Link System","text":""},{"location":"adv-topics/audio-level/#introduction","title":"Introduction","text":"

    Proper audio level calibration is crucial for ensuring good audio quality, and consistent audio levels from node-to-node. Fortunately, the procedure is not that complicated. To properly perform the audio level calibration, you will need the following items:

    "},{"location":"adv-topics/audio-level/#assumptions","title":"Assumptions","text":""},{"location":"adv-topics/audio-level/#procedure","title":"Procedure","text":""},{"location":"adv-topics/audio-level/#checking-the-radios-transmitters-peak-deviation","title":"Checking the Radio's Transmitter's Peak Deviation","text":"

    If you intend to drive the transmitter at the microphone input, or at another audio input before pre-emphasis and limiting, you must ensure that the peak deviation of the transmitter is no more than what is required to operate on the channel (typically 5KHz). If you driving the modulator directly from a URI, this section can be skipped.

    1. Disable the generation of CTCSS tones in the transmitter. (If you don't encode CTCSS, skip this step and steps 6 and 7)
    2. Hook up the audio signal generator to the transmitter microphone input, set it for a 1kHz sine wave, and the lowest output setting. Hook up the service monitor to the transmitter RF output (with appropriate attenuation for protection of your Service Monitor) and key the transmitter.
    3. On the service monitor, monitor the peak deviation of the transmitter. Gradually increase the output of the audio signal generator until further adjustment does not increase the deviation. Make a note of the peak deviation at this point.
    4. If the peak deviation is more than 5kHz, then adjust the transmitter deviation pot or soft pot so that the peak deviation reading on the service monitor is no more than 5kHz. Check the shape of the audio waveform on the service monitor's scope. It should show evidence of hard limiting (it should look like a square wave with rounded edges)
    5. Reduce the level of the audio signal generator so that the service monitor reads 3kHz of peak deviation. The audio waveform should now be a sine wave with no evidence of limiting.
    6. Re-enable the generation of CTCSS tones in the transmitter
    7. Check that the peak deviation is no more than 3.6 kHz. If it is more than this, adjust the CTCSS level so that a reading of 3.6kHz is attained.
    "},{"location":"adv-topics/audio-level/#adjusting-the-transmit-audio-level","title":"Adjusting the Transmit Audio Level","text":"
    1. Disable CTCSS tone generation in the transmitter or at the CTCSS encoder.
    2. Enable the test tone generator in rpt.conf and restart Asterisk
    3. Connect the radio interface to the transmitter
    4. Bring up the Asterisk CLI.
    5. From the asterisk CLI, type rpt fun yournodenumber *989 where yournodenumber is the node number assigned to you.
    6. The transmitter should key and you may hear a tone. Adjust the TX level until the service monitor indicates 3kHz of peak deviation.
    7. From the asterisk CLI, type rpt fun yournodenumber #. This will kill the test tone and unkey the transmitter.
    8. If encoding CTCSS, re-enable the CTCSS encoder in the transmitter, then repeat the previous 3 steps. You should see no more than 3.6KHz of deviation with the CTCSS tone generator enabled. If it is higher than this, adjust the CTCSS level per your transmitter's service documentation until a 3.6KHz of deviation is seen on the service monitor.
    "},{"location":"adv-topics/audio-level/#adjusting-the-receive-audio-level","title":"Adjusting the Receive Audio Level","text":"

    The procedure depends on what type of radio you have interfaced. If your radio is a repeater, use the full duplex procedure, if your radio is a remote base or simplex node, use the half-duplex procedure. Note: if you have a URI, the automated setup procedure should have set the receive level very close to what it needs to be. This section can be used to verify the receive level is correct for a URI, but the URI receive level should always be set using the automated adjustment procedure.

    "},{"location":"adv-topics/audio-level/#full-duplex-radios","title":"Full Duplex Radios","text":"

    Adjusting the receive level of a full duplex radio is easy.

    1. Disable the transmitter CTCSS encoder. Bypass the receiver CTCSS decoder so that it is in carrier squelch.
    2. Monitor the deviation of the transmitter on the service monitor while simultaneously injecting a 1kHz tone frequency at 3kHz deviation on the receiver.
    3. Adjust the receive level pot until you see 3kHz of deviation on the transmit frequency.
    4. Disable the signal on the receiver. The receive audio level is now set.
    "},{"location":"adv-topics/audio-level/#half-duplex-radios","title":"Half Duplex Radios","text":"

    Half duplex radios will require the use of a second node which has had the transmit level properly set, and is not transmitting a CTCSS tone. The second node must be receivable by your service monitor or deviation meter but does not need to be co-located with the node undergoing the receive level adjustment. The CODEC used to send audio to the second node must either be uLAW or ADPCM. The GSM codec will give inaccurate results.

    1. Using the Asterisk CLI, connect the node to be adjusted to the transmit node: rpt fun yournodenum *3txnodenum
    2. Monitor the deviation of the second node's transmitter on the service monitor while simultaneously injecting a 1kHz tone frequency at 3kHz deviation on the receiver on the first node.
    3. Adjust the receive level pot until you see 3kHz of deviation on the transmit frequency.
    4. Disable the signal on the receiver. The receive audio level is now set.
    5. Disconnect the receive node from the transmitter: rpt fun yournodenum *1txnodenum
    "},{"location":"adv-topics/autopatch/","title":"Autopatch","text":"

    The autopatch feature in app_rpt allows users on the radio to interconnect with the public switched telephone network.

    "},{"location":"adv-topics/autopatch/#regulatory-issues","title":"Regulatory Issues","text":"

    Some countries do not allow third-party traffic through autopatching or telephone interconnection on amateur radio frequencies. In addition, if a system is linked between one country which allows autopatching and one which doesn't, just the passing of the traffic itself across the link could be considered a violation of the rules in the prohibiting country. In countries which permit autopatching, users need be made aware of this and should take the node off-link before using the autopatch.

    "},{"location":"adv-topics/autopatch/#security-issues","title":"Security Issues","text":"

    The examples provided here do not secure the autopatch against toll fraud. A lot can be done to prevent or reduce toll fraud, but the prevention measures vary by where you are located geographically. Most of the prevention measures will be in extensions.conf. You can write contexts to reject certain number sequences. The asterisk book should be consulted to see how to do this, Before placing any autopatch in service, please make sure you have secured your system so that toll fraud does not become an issue.

    "},{"location":"adv-topics/autopatch/#selecting-a-voip-provider","title":"Selecting a VOIP Provider","text":"

    Call termination into the public switched telephone network is a service offered by a Voice Over IP provider (VOIP). This is a highly competitive business, and there are lots of VOIP providers offering termination for Asterisk users. Termination is usually provided using the SIP protocol, while IAX is offered by a few providers.

    When selecting an VOIP provider, make sure they provide setup instructions with example configurations including usernames, and passwords. Quite a few VOIPs will automatically generate a custom SIP or IAX stanza for insertion into sip.conf, or iax.conf respectively.

    "},{"location":"adv-topics/autopatch/#configuration-files","title":"Configuration Files","text":"

    Four configuration files will be used to setup the autopatch: rpt.conf, extensions.conf, modules.conf, and iax.conf or sip.conf. The VOIP provider should provide a stanza for you to insert in iax.conf or sip.conf. The configuration below shows how each of these files are used when an outgoing autopatch call is made.

    In the modules.conf file, add these modules at the bottom of the file, but above the [global] stanza:

    load => bridge_builtin_features.so\nload => bridge_builtin_interval_features.so\nload => bridge_holding.so\nload => bridge_native_rtp.so\nload => bridge_simple.so\nload => bridge_softmix.so\nload => chan_bridge_media.so\n

    In the rpt.conf configuration file, a context in extensions.conf is defined in a node stanza using the context key:

    [1234]\n\ncontext = autopatch                 ; dialing context for phone\ncallerid = \"Repeater\" <0000000000>  ; callerid for phone calls\naccountcode = RADIO                 ; account code (optional)\n
    Also in the rpt.conf file, entries are added to the function stanza so that users can bring up or take down the autopatch: 
    [functions]\n61 = autopatchup,noct=1,farenddisconnect=1,dialtime=20000,context=autopatch\n62 = autopatchdn \n

    In extensions.conf, a stanza for the autopatch context is added which refers to the peer stanza in iax.conf or sip.conf.

    For outbound IAX connections extensions.conf 

    [autopatch] \n\nexten => _1NXXNXXXXXX,1,Dial,IAX2/peername/${EXTEN} \nexten => _1NXXNXXXXXX,2,Congestion \n
    For outbound SIP connections it should look like this: 
    [autopatch] \n\nexten => _1NXXNXXXXXX,1,Dial,SIP/peername/${EXTEN} \nexten => _1NXXNXXXXXX,2,Congestion \n

    The peer stanza you insert in iax.conf or sip.conf should preferably be provided by your VOIP provider. You'll want to use the stanza they identify for performing outgoing connections. The SIP module has been removed from ASL3 and one must now use PJSIP. A nonoperational example of an iax2 peer stanza looks like this: 

    [peername]\ntype=peer \nhost=127.0.0.1 \nsecret=nunya\nauth=md5 \ndisallow=all \nallow=gsm \nallow=ulaw\n

    "},{"location":"adv-topics/autopatch/#autopatch-options","title":"Autopatch Options","text":"

    There are several options that can be passed to the autopatchup command class.

    This table summarizes what they do:

    Autopatch Option Description context Override the context specified for the autopatch in rpt.conf dialtime The maximum time to wait between DTMF digits when a telephone number is being dialed. The patch will automatically disconnect if this time is exceeded. The value is specified in milliseconds. farenddisconnect When set to 1, the patch will automatically disconnect when the called party hangs up. The default is send a circuit busy tone until the radio user brings the patch down. noct When this is set to 1, the courtesy tone during an autopatch call will be disabled. The default is to send the courtesy tone whenever the radio user unkeys. quiet When set to 1, do not send dial tone, voice responses, just try to connect the call. voxalways When set to 1, enables vox mode. exten Overrides the default autopatch extension. nostar Disables the repeater function prefix"},{"location":"adv-topics/autopatch/#simplex-autopatch","title":"Simplex Autopatch","text":"

    For nodes with duplex set to 0 or 1, Simplex Autopatch operation is supported. This is accomplished by using a VOX on the audio coming from the telephone call, because of the half-duplex nature of the node. To avoid having audio from the telephone call keep the transmitter engaged for extended periods of time (if there is some source of continuous audio), there are two time-out values.

    Vox timeout, which is the maximum amount of time that the VOX can hold the transmitter engaged, which by default is 10 seconds. The other one is the Vox recovery time, which is the amount of time that the Vox is disabled during times of continuous audio, after the Vox timeout, which is 2 seconds. During continuous audio, it transmits for 10 seconds, then stops for 2 seconds (so that you can transmit to it and perhaps disconnect the call), then transmits for another 10 seconds, then stops for 2 seconds. This repeats until the call is disconnected.

    Additionally, to prevent 'chop-off' of the first syllable or two of the audio from the telephone call, the audio is delayed to allow for the transmitter to start transmitting and any CTCSS tones to be decoded. This delay is typically 500ms, but can be adjusted using the 'simplexpatchdelay' parameter. It is specified in units of 20 milliseconds.

    These time-out values may be overridden by using the following configuration parameters located in the [node-number] section of rpt.conf: 

    [1999]\nvoxtimeout = 10000 ; vox timeout time in ms\nvoxrecover = 2000  ; vox recover time in ms\nsimplexpatchdelay = 25 ; Delay for transmit while in patch in 20ms increments\nsimplexphonedelay = 25 ; Delay for phone while in patch for 20ms increments\n

    "},{"location":"adv-topics/broadcastify/","title":"Streaming Node Audio to Broadcastify","text":"

    To broadcast your node's audio on Broadcastify, you will need a Broadcastify account. You can then apply for a feed. This link provides information on applying for a feed.

    After you have your account and feed credentials, you a ready to setup AllStar.

    "},{"location":"adv-topics/broadcastify/#setup-a-feed","title":"Setup a Feed","text":"

    The above parameters have these meanings: 

    -- preset cbr 16` = use constant bit rate 16\n-r = Assume the input file is raw pcm\n-m m = Mode mono\n-s 8 = sample rate 8\n--bitwidth 16 = bit width is 16 (default)   \n

    If you experience any problems, look at /var/log/ezstream.log for error messages.

    "},{"location":"adv-topics/broadcastify/#migrating-an-existing-feed","title":"Migrating an existing feed","text":"

    If you have an existing feed, you will need to upgrade your existing xml file to the new format. You can use the following commands. 

    cd /etc\nezstream-cfgmigrate -0 ezstream.xml > ezstream.xml.new\ncp ezstream.xml ~/\nmv ezstream.new ezstream.xml\n

    "},{"location":"adv-topics/cloud-kernel/","title":"Using AllStarLink/app_rpt with Cloud Kernels","text":"

    Many virtual hosts or VPS providers leverage the Debian \"cloud kernel\" version of the Debian Linux OS. These providers include Amazon AWS, Microsoft Azure, Linode, Vultr, DreamHost, and DigitalOcean. Using their stock Debian 12 installation image comes with the \"cloud kernel\" installed. While this kernel is usually beneficial due to its smaller footprint than the full kernel through the removal of hardware elements, the Asterisk app_rpt module requires the support of these elements.

    "},{"location":"adv-topics/cloud-kernel/#diagnosing-a-dahdi-kernel-issue","title":"Diagnosing a DAHDI Kernel Issue","text":"

    Problems with Asterisk/app_rpt related to the DAHDI module being unavailable can be see by the following log messages at boot:

    [2024-12-10 15:09:19.780] ERROR[930155] loader.c: res_timing_dahdi declined to load.\n[2024-12-10 15:09:19.818] NOTICE[930201] app_rpt.c: Normal Repeater Init  63001\n[2024-12-10 15:09:19.818] WARNING[930220][C-00000001] chan_dahdi.c: Unable to open '/dev/dahdi/pseudo': No such file or directory\n[2024-12-10 15:09:19.818] ERROR[930220][C-00000001] chan_dahdi.c: Unable to dup channel: No such file or directory\n[2024-12-10 15:09:19.818] ERROR[930220] app_rpt/rpt_bridging.c: Failed to request dahdi/pseudo\n[2024-12-10 15:09:19.818] WARNING[930220] app_rpt.c: Disabled broken repeater 63001\n

    This appears in /var/log/asterisk/messages.log not long after Asterisk begins to start.

    "},{"location":"adv-topics/cloud-kernel/#converting-from-cloud-to-normal-kernel","title":"Converting from Cloud to Normal Kernel","text":"

    After installing a Debian Linux 12 image that is using a cloud kernel, the following procedure can be used to change the OS to use the normal kernel that will allow the building of the DAHDI module needed for app_rpt.

    Now setup of ASL3/app_rpt should be able to continue.

    "},{"location":"adv-topics/commands/","title":"New Commands","text":"

    New and different ASL3 commands are mentioned here. Older ASL commands are still documented on the AllStarLink Wiki

    "},{"location":"adv-topics/commands/#app_rpt-commands","title":"app_rpt commands","text":" 
    [general]\nnode_lookup_method = both   ;method used to lookup nodes\n                            ;both = dns lookup first, followed by external file (default)\n                            ;dns = dns lookup only\n                            ;file = external file lookup only\n
    "},{"location":"adv-topics/commands/#chan_simpleusb-commands","title":"chan_simpleusb commands","text":"

    susb show settings is used to show the currently selected node's settings. This replaces susb tune.

    "},{"location":"adv-topics/commands/#chan_usbradio-commands","title":"chan_usbradio commands","text":"

    radio show settings is used to show the currently selected node's settings. This replaces radio tune.

    "},{"location":"adv-topics/commands/#simpleusb-and-usbradio-tune-menus","title":"SimpleUSB and USBRadio Tune Menus","text":"

    The simpleusb-tune-menu and radio-tune-menu utility programs have been updated with new options. The new options allow you change the operation of the respective channel driver, in realtime, without having to manually edit the simpleusb.conf or usbradio.conf files.

    In addition to the updating settings, you can now view the live status of the COS, CTCSS inputs and PTT output. This allows you to easily view and change their settings.

    "},{"location":"adv-topics/commands/#echolink","title":"Echolink","text":""},{"location":"adv-topics/commands/#echolink-commands","title":"Echolink commands","text":"

    echolink show nodes is used to view the currently connected echolink users.

    echolink show stats is used to view the channel statistics for echolink. It shows the number of in-bound and out-bound connections. It also shows the cumulative system statistics, along with the statistics for each connected nodes.

    "},{"location":"adv-topics/commands/#helper-scripts","title":"Helper scripts","text":""},{"location":"adv-topics/commands/#debugging","title":"Debugging","text":"

    Previously app_rpt and associated channels supported setting the debug level with an associated app / channel command. These app / channel commands have been removed and replaced with the asterisk command:

    core set debug x module Where x is the debug level and module is the name of the app or module.

    Examples:

    "},{"location":"adv-topics/commands/#usb-eeprom-operation","title":"USB EEPROM Operation","text":"

    chan_simpleusb and chan_usbradio allow users to store configuration information in the EEPROM attached to their interface(s). The CM119A can have manufacturer information in the same area that stores the user configuration. The CM119B does have manufacturer data in the area that stores user configuration. The manufacturer data cannot be overwritten. The user configuration data has been moved higher in memory to prevent overwriting the manufacturer data. If you use the EEPROM to store configuration data, you will need to save it to the EEPROM after upgrading. Use susb tune save or radio tune save.

    "},{"location":"adv-topics/conftmpl/","title":"Asterisk Templates","text":"

    Asterisk now has a template configuration option. This can make editing Asterisk conf flies less tedious. So far, ASL3 is using templated configurations in the \"rpt.conf\", \"simpleusb.conf\", and \"usbradio.conf\" files.

    "},{"location":"adv-topics/conftmpl/#etcasteriskrptconf","title":"/etc/asterisk/rpt.conf","text":"

    The \"rpt.conf\" file remains basically the same as it was in ASL2. But, the organization has changed to make it easier to maintain, particularly for multi-node systems.

    The template for a node in rpt.conf is named \"[node-main]\". Every node tagged with \"(node-main)\" inherits all the template settings. Template settings are overwritten by nodes with \"(main-node)\" attached.

    As you will see, adding a new node requires the addition of just a few lines. For example adding three nodes to your system may need nothing more than:

    [1999](node-main)\nrxchannel = SimpleUSB/1999\nidrecording = |iWB6NIL\nstartup_macro = *32000\n\n[1998](node-main)\n;this node would use the ID in the template\nrxchannel = Radio/1998\nmorse = morse_1998\n\n[1997](node-main)\n;This might be a hub node if \"rxchannel = dahdi/pseudo\" is in the template\n
    "},{"location":"adv-topics/conftmpl/#etcasterisksimpleusbconf-etcasteriskusbradioconf","title":"/etc/asterisk/simpleusb.conf, /etc/asterisk/usbradio.conf","text":"

    The \"/etc/asterisk/simpleusb.conf\" and \"/etc/asterisk/usbradio.conf\" files have also adopted a templated configuration. Again, the template for a node is named \"[node-main]\". Every node tagged with \"(node-main)\" inherits all the template settings. Template settings are overwritten by nodes with \"(main-node)\" attached.

    "},{"location":"adv-topics/conftmpl/#asl-menu-support","title":"ASL menu support","text":"

    The new ASL3 menu is fully aware of the templated configuration and handles adding, updating, and removing nodes.

    "},{"location":"adv-topics/hotspot_hd/","title":"Hot Spot Unkey Delay","text":"

    When setting up a hot spot select the Duplex 1 option with the asl-menu will allow the hot spot to function. However the unkey delay of the radio transmiter will be too long.

    These steps will shorting the unkey delay and will greatly improve hot spot responsiveness.

    This setup will likely be a single menu selection with the next release of ASL3.

    "},{"location":"adv-topics/httpreg/","title":"HTTP Registration","text":"

    AllStarLink is moving from IAX2 to HTTP registration. ASL3 has a new module for http registration. ASL3 menu users need not be concerned about this change. The asl3-menu sets this for you.

    The conf new file is /etc/asterisk/rpt_http_registrations.conf. The setting values are the same as they were in IAX2 registration.

    [General]\n\n[registrations]\n;Change the register => line to match your assigned node number and password.\nregister => 1999:password@register.allstarlink.org    ; This must be changed to your node number, password\n

    While IAX registration still works, please do not register with both HTTP and IAX. That would result in unnecessary server load for no gain. The long term plan is to do away with IAX registration. HTTP registration also allows load balancing and other advantages not available with IAX. The ASL3 IAX module has been replaced with the Asterisk LTS IAX2 module for upstream compatibility.

    "},{"location":"adv-topics/iaxreg/","title":"IAX-Based Registration","text":"

    This documents how to use IAX-based registration instead of HTTP-based registration with ASL3.

    Do Not Do This \"Just Because\"

    Only convert your node to using IAX-based registration because you have a known problem using HTTP registration. Use of IAX registration is significantly higher server use than HTTP registration and 99+% of all users DO NOT have a reason to use it.

    "},{"location":"adv-topics/iaxreg/#reason-to-use-iax-based-registration","title":"Reason to use IAX-Based Registration","text":"

    The following are the only reasons to use IAX-based registration with ASL3:

    1. Outbound HTTP/HTTPS is blocked but you have outbound UDP port access available. For example, your node sits behind a proxy or other security or content filtering device that breaks ASLv3 registration.

    2. Your node is behind a connections that handle HTTP traffic differently than IAX traffic. A known example is AT&T and T-Mobile hotspots. These hotspots intercept/direct web (HTTP/HTTPS) traffic and route it differently from IAX traffic. In these use cases, the HTTP-based registration will register a proxy/accelerator IP address rather than the IP being used by the node.

    "},{"location":"adv-topics/iaxreg/#using-iax-registration-instead-of-http-registration","title":"Using IAX Registration Instead of HTTP Registration","text":"

    To change from HTTP to IAX registration, do the following as root (i.e. use sudo -s):

    1. Edit /etc/asterisk/modules.conf and change the line for res_rpt_http_registrations.so from \"load\" to \"noload\": 
      noload => res_rpt_http_registrations.so\n

    2. Edit /etc/asterisk/iax.conf and add a register => line under the [general] stanza. This can be copied from a functional rpt_http_registrations.conf file: 

      [general]\nregister => 63001:VerySecret@register.allstarlink.org\n

    3. Restart asterisk with systemctl restart asterisk

    4. Run the asterisk console with asterisk -r and test the IAX registration with the command iax2 show registry: 

      node63001*CLI> iax2 show registry\nHost                  dnsmgr  Username  Perceived                 Refresh  State\n34.105.111.212:4569   Y       63001     192.0.2.171:4569          180      Registered\n

      If it lists \"State\" as \"Registered\" you're all set.

    "},{"location":"adv-topics/incompatibles/","title":"Incompatibilities with Legacy Versions & Configs","text":"

    AllStarLink v3 has certain incompatible changes with older versions of the AllStarLink system.

    "},{"location":"adv-topics/incompatibles/#voterrtcm-default-port","title":"Voter/RTCM Default Port","text":"

    Modern installations of Asterisk runs as the unprivileged asterisk user rather than as root. Standard Linux convention prohibits non-root users to listen on a TCP port below 1024. The default port for Voters/RTCMs has been changed to 1667 when previously it was 667.

    If the Voter/RTCM port cannot be easily changed, then the following configuration can be made to the underlying operating system:

    echo net.ipv4.ip_unprivileged_port_start=667 > /etc/sysctl.d/aslport667.conf\nsysctl -p\nsystemctl restart asterisk\n

    If running the AllStarLink Pi Appliance (or another system with a firewall), inbound to port 667/UDP must be permitted. For directions on how to do this with the Pi Appliance see Managing the Firewall.

    "},{"location":"adv-topics/incompatibles/#usb-udev","title":"USB udev","text":"

    A udev rule is needed to allow Asterisk running as non-root access to the USB system. ASL3 systems installed from debs, apt install or images will already have this rule in place.

    /etc/udev/rules.d/90-asl3.rules\nSUBSYSTEM==\"usb\", ATTRS{idVendor}==\"0d8c\", GROUP=\"plugdev\", TAG+=\"uaccess\"\n

    The UDev subsystem must be reloaded and then the USB device re-inserted into the port. UDev is reloaded with the command

    udevadm control --reload\n

    Alternatively, reboot the system.

    "},{"location":"adv-topics/noderesolution/","title":"Node Resolution","text":"

    app_rpt has three ways to resolve a node number to an IP address.

    The above mechanisms are used to translate a node number into its respective public IP address. Connections are made directly between nodes using their public IP addresses.

    app_rpt first looks to see if the node is statically configured. If it is not found, the setting for node_lookup_method in the [general] stanza of rpt.conf determines the next step. The default setting is to perform a DNS lookup and if not found, attempt a lookup using the external file.

    The operation of this ASL3 feature is controlled by the following information in /etc/asterisk/rpt.conf.

    [general]\nnode_lookup_method = both   ;method used to lookup nodes\n                    ;both = dns lookup first, followed by external file (default)\n                    ;dns = dns lookup only\n                    ;file = external file lookup only\n
    The node lookup routines will output debug information showing the node lookups if the debug level is set to 4 or higher.

    Note: The AllStarLink network maintains security by requiring node owners to register with AllStarLink. Before a node is published to our directory it must successfully authenticate and register with our server. Related see HTTP Registration

    "},{"location":"adv-topics/noderesolution/#external-node-directory-file","title":"External Node Directory File","text":"

    app_rpt can use an external node directory file to perform node resolution. The asl3-update-nodeList service is used to periodically download the current list of authenticated nodes.

    The name of this file is configured in rpt.conf using the extnodefile key and value. By default the name of this file is /var/lib/asterisk/rpt_extnodes. Multiple files can be specified based on your requirements. The file names are separated with a comma.

    When a file based lookup is performed, app_rpt reloads the external file before searching for a match. This allows external scripts the opportunity to keep the file updated.

    Here is an example of the external file format:

    [extnodes]\n\n2000=radio@162.248.93.134:4569/2000,162.248.93.134\n2001=radio@162.248.93.134:4569/2001,162.248.93.134\n
    "},{"location":"adv-topics/noderesolution/#domain-name-service-dns","title":"Domain Name Service DNS","text":"

    AllStarLink now provides a DNS service for performing node resolution. DNS is an Internet standard and is more efficient than using a file base solution.

    When using DNS app_rpt must know the length of the longest node number. The current maximum is 6 digits. This number can be overridden in rpt.conf by adding the max_dns_node_length key and value to the [general] stanza.

    For more information on DNS see AllStarLink DNS Servers.

    "},{"location":"adv-topics/noderesolution/#statically-configured-node","title":"Statically Configured Node","text":"

    In some cases, it is desirable to statically define the node information. This is commonly used for private nodes or private networks.

    To configure a static node, enter the information for the node in rpt.conf under the [node] stanza. Here is an example configuration:

    [nodes]\n1001 = radio@192.168.0.1/1001,NONE\n1998 = radio@127.0.0.1:4568/1998,NONE\n1999 = radio@192.168.0.2:4569/1999,NONE\n
    "},{"location":"adv-topics/other-software/","title":"Other Software Products","text":"

    The following information relates to software components related to AllStarLink.

    "},{"location":"adv-topics/other-software/#web-dashboards","title":"Web Dashboards","text":"

    The primary web dashboard supported by AllStarLink is Allmon3.

    But, we know there are other dashboards available and everyone has their own personal favorite. We also know that some of the other dashboards make use of database (\"astdb.txt\"). Typically, this DB has been downloaded by a PHP script (\"astdb.php\") and updated with the help of boot scripts and cron jobs.

    To support these other dashboards we have included a systemd service to download and update the \"astdb.txt\" database. By default, this service and its associated timer are disabled. If you have installed a dashboard that is dependent on this database you will need to execute the following commands :

    systemctl enable asl3-update-astdb.service\nsystemctl enable asl3-update-astdb.timer\nsystemctl start asl3-update-astdb.timer\n
    "},{"location":"adv-topics/permissions/","title":"Permissions","text":""},{"location":"adv-topics/permissions/#asterisk-vs-root","title":"\"asterisk\" vs \"root\"","text":"

    One of the new features with ASL3 is that the Asterisk process now runs as the \"asterisk\" user (not as the super-user; \"root\"). While sometimes handy this was potentially very dangerous. This change was made for increased security. It's a good thing!

    We acknowledge that there are some applications, commands, and scripts that you may want to bring over from older (and other) versions of ASL and that these may be impacted by ASL3's heightened security. Porting these over to ASL3 may require some simple changes to filesystem permissions. Other issues may need other solutions (e.g. you can't execute a command that requires \"root\" privileges when you are not the \"root\" user). For guidance, reach out to the AllStarLink Community.

    In short, in ASL3 the Asterisk process now runs as the \"asterisk\" user. If you modify the \"/etc/asterisk/rpt.conf\" file to execute a command (or script) it will NOT have super-user (\"root\") privileges. The command itself and any file it references must be readable by the \"asterisk\" user. Any files that the command needs to modify must be writable by the \"asterisk\" user. Lastly, the parent directory of any file being created must also be writable.

    Asterisk User and Sudo

    Do not give general access to sudo to the asterisk user. This is very dangerous and poor practice. While people will state \"I've always done it this way\" it is, in fact, a serious security risk. As AllStarLink is run on the public internet for many repeater sites, the development team prioritizes security of the system. For common uses of this, namely restarting services and shutting down a node, see Managing Services and OS Shutdowns/Reboots.

    "},{"location":"adv-topics/permissions/#filesystem-permissions","title":"Filesystem Permissions","text":"

    The following is a very brief overview of Linux filesystem permissions. More in-depth information is available on the internet (search for \u201clinux file permissions\u201d).

    On a Linux system, each file (and directory) has permissions that effect access by the \u201cowner\u201d, permissions that effect access to those logins in the same \u201cgroup\u201d, and permissions that effect access to \u201cother\u201d logins. The permissions determine whether you (or some other process) can read, write, or execute the files (or search in directories). There are commands to change the ownership (\u201cchown\u201d) and permissions (\u201cchmod\u201d) for each file/directory.

    "},{"location":"adv-topics/permissions/#file-permissions","title":"File permissions","text":"

    You can use the ls -l [FILE] command to view file permissions. The permissions determine whether the calling process can read (\"r\"), write (\"w\"), or execute (\"x\") the file. Here are some examples :

    node63001:~/examples $ ls -l *file*\n-rw------- 1 root root     0 Aug  5 08:28 1-file-owner-root-can-read-write\n-rw-r----- 1 root asterisk 0 Aug  5 08:28 2-file-add-group-asterisk-can-read\n-rw-r--r-- 1 root asterisk 0 Aug  5 08:27 3-file-add-everyone-can-read\n\nnode63001:~/examples $ ls -l *script*\n-rwx------ 1 root root     0 Aug  5 08:28 1-script-owner-root-can-read-write-execute\n-rwxr-x--- 1 root asterisk 0 Aug  5 08:28 2-script-add-group-asterisk-can-read-execute\n-rwxr-xr-x 1 root asterisk 0 Aug  5 08:27 3-script-add-everyone-can-read-execute\n

    In each case, the file permissions progress from the most restrictive (only the file owner can read/execute) to most available (all users can read/execute).

    "},{"location":"adv-topics/permissions/#directory-permissions","title":"Directory permissions","text":"

    You can use the ls -ld <directory> command to view the permissions on a directory (vs. the contents of a directory). The permissions determine whether the calling process can read (\"r\"), write (\"w\"), or search (\"x\") the directory. Here are some examples :

    node63001:~/examples $ ls -ld *dir*\ndrwx------ 2 root root     4096 Aug  5 09:56 1-dir-owner-root-can-read-write-search\ndrwxr-x--- 2 root asterisk 4096 Aug  5 09:56 2-dir-add-group-asterisk-can-read-search\ndrwxr-xr-x 2 root asterisk 4096 Aug  5 09:56 3-dir-add-everyone-can-read-search\n

    In each case, the directory permissions progress from the most restrictive (only the directory owner can read/write/search) to most available (all users can read/search).

    "},{"location":"adv-topics/permissions/#changing-permissions","title":"Changing permissions","text":"

    The following commands can be used to change the permissions of files and directories :

    Command Description Sample Usage chown change file owner and group # chown root /var/asl-backups/asl-backup-files chgrp change group ownership # chgrp asterisk /var/spool/asterisk/monitor chmod change file mode bits # chmod 640 /etc/asterisk/manager.conf

    Any process with \"write\" permission to a file can update (including completely overwrite) the file. A process with \"write\" permission to a directory can add/remove/rename files in that directory. If a command/script is writable then it can be changed to \"do something different\" when it is next executed. When updating filesystem permissions, please remember that doing so can potentially expose the contents of files (and directories) to others. Be cautious!

    "},{"location":"adv-topics/permissions/#managing-services-and-os-shutdownsreboots","title":"Managing Services and OS Shutdowns/Reboots","text":"

    Beginning with asl3-3.4.0, AllStarLink v3 comes with a PolicyKit ruleset to permit the asterisk user to execute a limited number of actions without the need for sudo or prompting for a password. Those are:

    Using a combination of wrapper scripts and appropriate function configuration, Asterisk can restart itself, Allmon3, shutdown the system, or reboot the system. Use of the wrapper scripts for systemctl commands is essential for a clean execution of the scripts under the polkit rules. The provided wrappers are:

    These privileges can be used inside /etc/asterisk/rpt.conf within the [functions] stanza as follows:

    9001 = cmd,/etc/asterisk/scripts/asterisk-restart\n9002 = cmd,/etc/asterisk/scripts/asterisk-stop\n9003 = cmd,/etc/asterisk/scripts/allmon3-restart\n9004 = cmd,/etc/asterisk/scripts/allmon3-stop\n9005 = cmd,/usr/sbin/shutdown\n9006 = cmd,/usr/sbin/reboot\n

    The command *9001 would restart asterisk, *9006 would reboot the system, etc.

    "},{"location":"adv-topics/sa818modules/","title":"SA818 RF Modules","text":"

    Setting up an SA818 RF module is much easier on ASL3.

    Two new commands were added to ASL3 Pi Appliance systems to program the module. We believe that most will prefer using sa818-menu, a visual interface to the RF module settings. But, for those who prefer a command line interface we have also included sa818.

    "},{"location":"adv-topics/sa818modules/#sa818-menu","title":"sa818-menu","text":"

    To program your SA818 with the menu executing the following command from the Linux shell :

    sa818-menu\n

    Here, you will be able to view the last saved (or default) settings. Please note that the \"last saved\" settings are those saved by the sa818-menu command on your system. We have no way to \"read\" settings from the SA818 RF module.

    The settings include :

    Setting Description Band Specify the module type (VHF, UHF) Bandwidth Narrow (12.5KHz) or Wide (25.0KHz) Receive Frequency Transmit Frequency Squelch Value Default = 1 Volume Default = 1 Sub-audible tone None, CTCSS, DCS Pre-Emphasis/De-emphasis Default = Disabled High pass Filter Default = Disabled Low pass Filter Default = Disabled Serial Port Default = /dev/serial0 or /dev/ttyUSB0 Connection Speed Default = 9600

    If you change the sub-audible tone setting to CTCSS or DCS you will be presented with a list of possible tone frequencies. When using CTCSS tones, you will also have a setting for the CTCSS Reverse Burst (Tail Tone).

    The RF module will be programmed when you exit the menu.

    Note: if you make changes that you do not wish to have stored then use the ESC key to exit the menu.

    "},{"location":"adv-topics/sa818modules/#sa818","title":"sa818","text":"

    To program your SA818 using the command line you can execute the sa818 command. For command usage information, use sa818 --help.

    "},{"location":"adv-topics/sip-phone/","title":"Setting up a SIP Phone","text":"

    This document describes the steps necessary to set up a SIP phone in ASL3. The setup procedure has changed due to the deprecation of chan_sip. Users are now required to use chan_pjsip. chan_pjsip brings new features to the SIP stack and is the supported SIP channel for the future.

    "},{"location":"adv-topics/sip-phone/#update-modulesconf","title":"Update modules.conf","text":"

    chan_pjsip requires a number of modules to be loaded. You should start by editing /etc/asterisk/modules.conf and add the following to the [modules] stanza (generally near the bottom of the file just above the [global] stanza):

    ;\n; modules for pjsip\n;\nnoload = app_voicemail.so\nload = bridge_builtin_features.so\nload = bridge_builtin_interval_features.so\nload = bridge_holding.so\nload = bridge_native_rtp.so\nload = bridge_simple.so\nload = bridge_softmix.so\nload = chan_bridge_media.so\nload = chan_pjsip.so\nload = func_pjsip_endpoint.so\nload = func_sorcery.so\nload = func_devstate.so\nload = res_pjproject.so\nload = res_pjsip_acl.so\nload = res_pjsip_authenticator_digest.so\nload = res_pjsip_caller_id.so\nload = res_pjsip_dialog_info_body_generator.so\nload = res_pjsip_diversion.so\nload = res_pjsip_dtmf_info.so\nload = res_pjsip_endpoint_identifier_anonymous.so\nload = res_pjsip_endpoint_identifier_ip.so\nload = res_pjsip_endpoint_identifier_user.so\nload = res_pjsip_exten_state.so\nload = res_pjsip_header_funcs.so\nload = res_pjsip_logger.so\nload = res_pjsip_messaging.so\nload = res_pjsip_mwi_body_generator.so\nload = res_pjsip_mwi.so\nload = res_pjsip_nat.so\nload = res_pjsip_notify.so\nload = res_pjsip_one_touch_record_info.so\nload = res_pjsip_outbound_authenticator_digest.so\nload = res_pjsip_outbound_publish.so\nload = res_pjsip_outbound_registration.so\nload = res_pjsip_path.so\nload = res_pjsip_pidf_body_generator.so\nload = res_pjsip_pidf_digium_body_supplement.so\nload = res_pjsip_pidf_eyebeam_body_supplement.so\nload = res_pjsip_publish_asterisk.so\nload = res_pjsip_pubsub.so\nload = res_pjsip_refer.so\nload = res_pjsip_registrar.so\nload = res_pjsip_rfc3326.so\nload = res_pjsip_sdp_rtp.so\nload = res_pjsip_send_to_voicemail.so\nload = res_pjsip_session.so\nload = res_pjsip.so\nnoload = res_pjsip_t38.so\nnoload = res_pjsip_transport_websocket.so\nload = res_pjsip_xpidf_body_generator.so\nload = res_rtp_asterisk.so\nload = res_sorcery_astdb.so\nload = res_sorcery_config.so\nload = res_sorcery_memory.so\nload = res_sorcery_realtime.so\n
    "},{"location":"adv-topics/sip-phone/#update-extensionsconf","title":"Update extensions.conf","text":"

    The next step is to update extension.conf. This creates a dial plan that controls access to your node by SIP phones. Edit /etc/asterisk/extensions.conf and add the following to the bottom of the file. This example will set up extension 1001 as your local phone and the dialing plan for your node.

    [sip-phones]\nexten => 1001,1,Dial(PJSIP/${EXTEN},60,rT)\n\nexten => ${NODE},1,Ringing\nexten => ${NODE},n,Answer(3000)\nexten => ${NODE},n,Set(NODENUM=${CALLERID(number)})\nexten => ${NODE},n,Playback(extension)\nexten => ${NODE},n,SayDigits(${NODENUM})\nexten => ${NODE},n,Playback(rpt/connected)\nexten => ${NODE},n,Playback(rpt/node)\nexten => ${NODE},n,SayDigits(${EXTEN})\nexten => ${NODE},n,rpt(${EXTEN}|P)\nexten => ${NODE},n,Hangup\n

    The above dial plan will allow your SIP phone to dial your node number. In this example one should use *99 to enable PTT and # to disable PTT from the SIP phone. VOX is also an option for phone connections. When using VOX, one may want to mute the mic on the phone when not transmitting and unmute to transmit. VOX is enable by adding a 'v' to the rpt app line like this:

    exten => ${NODE},n,rpt(${EXTEN}|Pv)\n

    This dial plan announces the extension number and the connecting node number. The lines with SayDigits and Playback can be removed if you do not want the announcement.

    The ${NODE} variable is defined at the top of extensions.conf and should be your assigned node number. See the following.

    [globals]\nHOMENPA = 999 ; change this to your Area Code\nNODE = 1999   ; change this to your node number\n

    The entry 1001 in extensions.conf is the extension number for your SIP phone. If you want to use a different extension number, change 1001 to your desired extension number.

    Note: If you attempt to copy extensions.conf from a previous release of AllStarLink, it will fail. AllStarLink version 3 requires comma delimiters instead of pipe symbols for standard asterisk functions. The rpt function continues to use a pipe delimiter. This is subject to change in a future release.

    "},{"location":"adv-topics/sip-phone/#update-pjsipconf","title":"Update pjsip.conf","text":"

    You are now ready to configure pjsip. Edit /etc/asterisk/pjsip.conf. Don\u2019t get overwhelmed by all of the text in the sample configuration. pjsip supports several configurations. This document focuses on the necessary entries to get a SIP phone operational.

    Scroll down to the Basic UDP transport section. It should be updated to look like the following:

    ; Basic UDP transport\n;\n[transport-udp]\ntype=transport\nprotocol=udp    ;udp,tcp,tls,ws,wss,flow\nbind=0.0.0.0:5060\n

    Note: After the bind port address (0.0.0.0) is the incoming port number (5060). This UDP port must not be blocked by your network firewall.

    Scroll down to the section titled Endpoint Configured For Use With A Sip Phone. For each extension, you will need to enter 3 sections. Here is an example for extension 1001.

    [1001]\ntype=endpoint\ntransport=transport-udp\ncontext=sip-phones\ndisallow=all\nallow=ulaw\nallow=alaw\nallow=gsm\nauth=1001\naors=1001\ncallerid=\"My CallerID\"\n;\n; A few more transports to pick from, and some related options below them.\n;\n;transport=transport-tls\n;media_encryption=sdes\n;transport=transport-udp-ipv6\n;transport=transport-udp-nat\n;direct_media=no\n;\n; MWI related options\n\n;aggregate_mwi=yes\n;mailboxes=6001@default,7001@default\n;mwi_from_user=6001\n;\n; Extension and Device state options\n;\n;device_state_busy_at=1\n;allow_subscribe=yes\n;sub_min_expiry=30\n;\n; STIR/SHAKEN support.\n;\n;stir_shaken=no\n;stir_shaken_profile=my_profile\n\n[1001]\ntype=auth\nauth_type=userpass\npassword=1001\nusername=1001\n\n[1001]\ntype=aor\nmax_contacts=4\n;contact=sip:6001@192.0.2.1:5060\n

    If you want to use a different extension number, you will need to change [1001] with your desired number. The item callerid=\"My CallerID\" should also be updated for your preferred caller identifier.

    The items password=1001 and username=1001 should be updated to match the username and password you want for your extension number. You should use a complex password. Do not use the extension number. You can use the password generator located here to generate a complex password.

    These three sections can be replicated for each extension that you want to add to the system.

    "},{"location":"adv-topics/sip-phone/#restart-asterisk","title":"Restart asterisk","text":"

    After completing these changes, you must restart asterisk or reboot your system.

    systemctl restart asterisk\n
    "},{"location":"adv-topics/sip-phone/#example-config-with-multiple-sip-phones","title":"Example config with multiple SIP phones","text":"

    As noted above you can have more SIP phones. Here is an example of a simple dial plan that has multiple extensions.

    [sip-phones]\n; Extension 210 - Line 1\n; Extension 211 - Line 2\n; Extension 212 - Garage\n; Extension 213 - Cordless ATA\n; Extension 1000 - Voice Mail\n\nexten => 210,1,Dial(PJSIP/210,60,rT)\nexten => 211,1,Dial(PJSIP/211,60,rT)\nexten => 212,1,Dial(PJSIP/212,60,rT)\nexten => 213,1,Dial(PJSIP/213,60,rT)\nexten => 1000,1,VoiceMailMain(210)\nexten => 1000,2,Hangup\n\n; Allow SIP calls to local nodes\nexten => 1998,1,rpt(1998|P)\nexten => 1999,1,rpt(1999|P)\n

    This example includes voicemail. To use voice mail, you will need to change noload = app_voicemail.soto load = app_voicemail.so.

    "},{"location":"adv-topics/sip-phone/#troubleshooting","title":"Troubleshooting","text":"

    If you have trouble connecting your SIP phone, start the asterisk command line with asterisk -rvvv. Enter the command pjsip set logger on. This will show the communications between the SIP phone and asterisk. It will also show the actions in your dial plan.

    "},{"location":"adv-topics/sip-phone/#security","title":"Security","text":"

    If you will be exposing your system to the outside world. You should consider using fail2ban to protect the system.

    "},{"location":"adv-topics/tts/","title":"ASL Text-to-Speech","text":"

    ASL provides the asl-tts command. This is a text to speech speaker for ASL3. It uses the piper open source TTS system that is not depending on cloud providers and has opensource voice models for dozens of languages.

    "},{"location":"adv-topics/tts/#basic-operation","title":"Basic Operation","text":"

    asl-tts uses the piper-tts text to speech engine to generate ulaw files from text so that Asterisk/apt_rpt can speak any set of arbitrary text without needing to have a sound file installed for the word necessary. By default, asl-tts will cause app_rpt to immediately speak whatever text was specified.

    asl-tts has a time delay

    Running the piper-tts system takes a few seconds to compile the voice into the sound file. Larger text blocks may take tens of seconds to compile. asl-tts is not appropriate for time-sensitive communications such as speaking the current time.

    The command format is as follows: 

    asl-tts -n NODE -t 'TEXT' [ -v VOICE ] [ -f FILE ]\n

    The general use case is to speak to a node immediately. This this case speaking the text \"Good morning\" to node 63001 would look like this:

    sudo asl-tts -n 63001 -t 'Good morning'\n

    The TTS engine will start, create the audio, and execute the playback on the node. The playback follows normal app_rpt rules and will be played back as soon as its turn arrives. The command will not preempt or \"talk over\" other traffic on the node.

    Keep in mind the differences between quoting in a shell command. In general, the \" and ' work the same by the double-quote will allow variable substitution while the single quote will not. This can be useful if you want to create text based on a system condition. For example:

    TODAY=$(date +\"%A, %B %d, %Y\")\nsudo asl-tts -n 63001 -t \"Today is ${TODAY}\"\n

    Obviously the above example is a bit contrived as the date command doesn't generate nice words such as \"first\", \"seventeenth\", etc. However it demonstrates the flexibility of the tool. A script could be used to populate $TODAY with natural words. For a hypothetical example:

    TODAY=$(/path/to/today-script)\nsudo asl-tts -n 63001 -t \"Today is ${TODAY}\"\n

    More complex message can be created with additional shell coding. As an example here's how the current public IP address of the system could be obtained and spoken:

    IPADDR=$(wget -q -O - checkip.dyndns.com | grep -Po \"[\\d\\.]+\" | sed 's/\\./ dot /g')\nMSG=\"The node public IP is ${IPADDR}\"\nsudo asl-tts -n 63001 -t ${MSG}\n

    A bash shell variable can generally hold up to a 2 Mb-sized string so any reasonably-formatted text can be stored in a variable and then played with the -t command. Keep in mind that the text or the variable contents must not contain the same quote type used in the command without escaping it with a backslash. For example \"Nested \\\" mark\".

    Temporary File Cleanups

    Note that temporary files are written to directories named /tmp/asl-tts and are not automatically cleaned up because it's impossible to know when Asterisk will actually need/speak the file contents. If this becomes a space problem, put in a systemd timer unit or a cron job to delete old files in /tmp/asl-tts. The command can simply be rm -rf /tmp/asl-tts*.

    "},{"location":"adv-topics/tts/#creating-a-sound-file-for-later","title":"Creating a Sound File for Later","text":"

    To create the text for other usage, such as cron jobs, use the \"-f FILE\" option to specify where the file should be created. Do not use an extension; the file will be automatically appended with the .ul postfix.

    For example: 

    AUD_TEXT=\"The club net will begin in five minutes!\"\nasl-tts -n 460181 -t \"${AUD_TEXT}\" -f /tmp/netstartsin5\n

    This will create a sound file /tmp/netstartsin5.ul that can be used later with an asterisk playback such as: 

    asterisk -rx 'rpt playback 63001 /tmp/netstartsin5\n

    Note that -n is required but irrelevant to this use case.

    "},{"location":"adv-topics/tts/#voices","title":"Voices","text":"

    By default, the package provides on voice \"Amy\" from https://github.com/rhasspy/piper/blob/master/VOICES.md. This voice is the closest voice to the default Asterisk and app_rpt sounds. It is possible to download other voices, for any language, and store the .onnx and .onxx.json files in /var/lib/piper-tts and then use them with the -v option. For example, to add the English en_GB voice alan from the voices repository:

    1. Download the .onnx and .onnx.json file to /var/lib/piper-tts

    2. Specify -v en_GB-alan-low.onnx on the command line to asl-tts. As an example:

    asl-tts -n 63001 -v en_GB-alan-low.onnx -t \"Good morning\"\n

    Note, since all files are squashed down to 8K uLaw format, there is no value in the \"medium\" or \"high\" quality models. Always use the low quality model.

    "},{"location":"adv-topics/tts/#reporting-bugs","title":"Reporting Bugs","text":"

    Report bugs to https://github.com/AllStarLink/asl3-tts/issues

    "},{"location":"adv-topics/uefi-secureboot/","title":"UEFI and SecureBoot","text":"

    UEFI (Unified Extensible Firmware Interface) is a modern firmware interface that replaces the older BIOS system, providing a more flexible and powerful environment for booting and managing hardware. It supports larger hard drives, faster boot times, and a more user-friendly interface.

    Secure Boot is a feature of UEFI that ensures only trusted software is loaded during the boot process, protecting the system from malware and unauthorized software. It works by verifying the digital signatures of boot loaders and operating system files against a database of trusted certificates.

    Together, UEFI and Secure Boot enhance the security and efficiency of modern computing systems.

    Given a quirk of how Asterisk + app_rpt needs a kernel module (driver) not already contained in the core system provided by Debian, installing AllStarLink on a system requiring UEFI/SecureBoot requires additional steps to create what's called a Machine Owner Key.

    Note

    Raspberry Pi does not use or support UEFI or SecureBoot

    "},{"location":"adv-topics/uefi-secureboot/#generating-a-machine-owner-key","title":"Generating a Machine Owner Key","text":"

    Generation of a Machine Owner Key (MOK) is can be done with the asl-setup-dkms-mok utility. This tool appeared in the package asl3-3.3.0-1.

    1. Run the command sudo asl-setup-dkms-mok. This will prompt for several items of information. An example run will look like:

      $ sudo asl-setup-dkms-mok\n!!!\n!!! WARNING - RUNNING THIS SCRIPT WILL REGENERATE THE\n!!! DEFAULT MACHINE OWNER KEY (MOK) FOR SECURE BOOT. IN GENERAL YOU\n!!! ARE RUNNING THIS SCRIPT BECAUSE YOU NEED TO AND YOU KNOW WHY\n!!! YOU ARE DOING IT. HOWEVER BE WARNED!\n!!!\nENTER \"galaxy\" TO CONTINUE>> galaxy\n\nGenerating an MOK requires a name for the certificate. This can be\nanything, however your name or your callsign is recommended.\n\nName or callsign>> N8EI\n\nGenerating the MOK...\n......+...+...+.....+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*................+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*.+.........+......+...+.+...........+...+.+...+..+.+...+..............+.......+...........+.........+................+..+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++\n.....+............+.........+.+.....+....+...+...+..............+.+..+...............+.+..+...+.+........+.+......+........+...............+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*.......+...+............+..+....+............+..+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*....+.....+...+...........................+......+.......+...............+.....+......+..........+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++\n-----\n\nEnter a one-time password for this key. This password will be prompted\nfor at the next reboot to properly install the key in the UEFI\nbootloader keyring. This does not need to be an incredibly complex\npassword and you may write it down until it's used.\n\ninput password:\ninput password again:\n\nSetup /etc/dkms/framework.conf.d/asl.conf\nGenerate /etc/dkms/asl_sign_helper.sh\n\nYou must reboot the system and enter the OTP to load the key into SecureBoot\nThis will appear on your directly-connected monitor or your cloud console\n

    2. Prepare to Reboot the System. Make sure a monitor is attached to the system or, if using a cloud virtual machine / VPS, make sure that a remote virtual console is connected. Then, reboot the system.

      MOK Load Staging is One-Time Only

      Note that the MOK is only staged for load one time. If the MOK load prompt is not responded to at the first/next reboot, the MOK staged for load is wiped and Step #1 will have to be run again.

    3. Complete the steps to review, unlock, and load the MOK

      The next step varies by vendor

      The next step uses screenshots from a Linux/QEMU virtual environment. Every device manufacturer or cloud provider will have a slightly different process or look.

    4. Reboot the system back into Linux and log in.

    5. Check that the MOK created is now loaded by looking for the \"name\" you provided the certificate in step 1. In this example, the MOK was named \"N8EI\".

      sudo dmesg | grep -i N8EI\n

      The output will look like: 

      $ sudo dmesg | grep -i N8EI\n[    0.674788] integrity: Loaded X.509 cert 'N8EI: 92ed492eeb11594eb25af5cc70ff0587115f2577'\n

    6. Rebuild the dahdi-dkms package so that the modules are signed by the MOK so that the kernel will load then.

      apt install --reinstall -y dahdi-dkms\n

    7. Test the module loads successfully:

       $ sudo modprobe dahdi dahd_dummy dahdi_transcode\n$ lsmod | grep dahdi\ndahdi_dummy            16384  0\ndahdi_transcode        16384  0\ndahdi                 258048  2 dahdi_dummy,dahdi_transcode\n
    "},{"location":"adv-topics/usbinterfaces/","title":"USB Audio Interfaces","text":"

    Setting up USB audio interfaces is much easier with ASL3.

    The ASL3 menu and Asterisk CLI USB config commands handle these changes.

    "},{"location":"adv-topics/usbinterfaces/#usb-interface-ic-support","title":"USB Interface IC Support","text":"

    Nearly all USB Interface products that are used with ASL nodes use C-Media CM108 and CM119 USB Interface ICs. These ICs provide the following features:

    The CM119 provides an additional 4 GPIO lines (GPIOs 5-8) vs. the CM108. The original CM108 and CM119 ICs were replaced in the early 2010's with the CM108AH and CM119A. These ICs are still used in many USB Radio Interfaces (URIs). Newer versions of the CM1xx ICs \u2013 the CM108B and CM119B are now available and have some minor differences in gain settings and other specifications but are fully compatible with the earlier IC versions and with ASL.

    "},{"location":"adv-topics/usbinterfaces/#setting-audio-levels","title":"Setting Audio Levels","text":"

    When setting up a new node, or if any changes have been made \u2013 such as use of a different radio, microphone, or other hardware or gain settings \u2013 checking and setting audio levels is very important. Radio equipment such as HTs or mobile radios, and many electronics products such as VOIP phones often restrict end-user audio adjustment capabilities to ensure consistent audio levels. Allstar nodes and USB interfaces however generally allow greater flexibility and thus extra attention is required to be sure your audio interface gains are properly matched to the external audio input and output levels. Otherwise your audio levels could be much quieter or louder than they should be which could be disruptive to other users and systems.

    Most nodes use the SimpleUSB channel driver. Nodes that require more advanced features such as DSP squelch detection, CTCSS tone encode/decode or additional audio filtering options will use the USBRadio channel driver. These channel drivers both have tune utilities for setting audio levels and other functions. Once a node has been configured in the asl-menu \"AllStar Node Setup Menu\", select the \"Interface Tune CLI\" menu option. The following options should then be used to check and optimize audio level settings:

    Note: Be sure you are not connected to any other nodes when doing any of the following tests and adjustments.

    The tune utilities can also be run from the command line without needing to use asl-menu. For SimpleUSB run \"sudo /usr/sbin/simpleusb-tune-menu\" or for USBRadio run \"sudo /usr/sbin/radio-tune-menu\".

    "},{"location":"adv-topics/usbinterfaces/#eeprom-operation","title":"EEPROM Operation","text":"

    The SimpleUSB and USBRadio channel drivers allow users to store configuration information in a EEPROM IC attached to the C-Media CM1xx USB Interface IC. The CM119A can have manufacturer information in the same area that stores the user configuration. The CM119B does have manufacturer data in the area that stores user configuration. The manufacturer data cannot be overwritten. The user configuration data has been moved higher in memory to prevent overwriting the manufacturer data. If you use the EEPROM to store configuration data, you'll need to save it to the EEPROM after upgrading. Use susb tune save or radio tune save.

    "},{"location":"allmon3/basics/","title":"Basics","text":"

    Allmon is the standard web-based monitoring and management for the AllStarLink v3. Allmon3 is the next generation of web dashboards for AllStarLink, rewritten for performance, scalability, and responsiveness.

    "},{"location":"allmon3/basics/#design-goals","title":"Design Goals","text":"

    Allmon3 features and functionality shall be governed by the following guidelines:

    "},{"location":"allmon3/basics/#community","title":"Community","text":"

    All code of Allmon3 not otherwise licensed (e.g., Bootstrap, etc.) is licensed under the GNU Affero General Public License v3. The choice of the AGPLv3 promotes giving back to the amateur radio and ASL communities.

    "},{"location":"allmon3/config/","title":"Configuration","text":"

    All configuration resides in /etc/allmon3. The stock configuration files are always available at /usr/share/doc/allmon3/ for recovery and documentation.

    After changing any configuration file, the service allmon3 must be restarted using systemctl restart allmon3.

    "},{"location":"allmon3/config/#node-configuration","title":"Node Configuration","text":"

    Allmon3's node connections are configured in /etc/allmon3/allmon3.ini. The allmon3.ini file is a standard INI-formatted file. Each stanza in the file is a node number. Each node stanza then can take the following options:

    Option Req'd / Opt Default Description host Req'd - DNS name or IP address of the Asterisk/ASL node port Opt 5038 Port of the Asterisk manager user Req'd - username of the Asterisk monitor, most commonly 'admin' pass Req'd - password of the monitor user multinodes Opt - this node is a server hosting multiple nodes and this is the \"primary\" record for the host. voters Opt n[,n,...] List voters on this server, comma separate. No value disabled voters pollinterval Opt 1 polling interval to asterisk in default is 1. this value can be expressed as a decimal fraction of a second - e.g., .5 is 500ms, .375 is 375ms, etc. vpollinterval Opt 1 Broadcast interval of voter data in seconds default is 1. this value can be expressed as a decimal fraction of a second - e.g., .5 is 500ms, .375 is 375ms, etc. retryinterval Opt 15 Seconds between retries if initial connection to asterisk is lost (optional, default 15) retrycount Opt \"Infinite\" number of times to retry a lost asterisk connection before ending (default infinite)

    An example minimal node configuration would be: 

    [12345]\nhost = 127.0.0.1\nuser = admin\npass = Passw0rd\n

    An example of a server with multiple nodes on it: 

    [12345]\nhost = 127.0.0.1\nuser = admin\npass = Passw0rd\nmultinodes = 12345, 23456\n

    Here's an example for monitoring three ASL Nodes: 

    [50815]\nhost=172.17.16.36\nuser=admin\npass=password\n\n[460180]\nhost=172.17.16.217\nuser=admin\npass=password\n\n[48496]\nhost=192.2.0.145\nuser=admin\npass=password\nvoter=y\nvotertitle=Megavoter\n

    "},{"location":"allmon3/config/#server-customization","title":"Server Customization","text":"

    Allmon3 has multiple configuration files to consider:

    "},{"location":"allmon3/config/#webini","title":"web.ini","text":"

    The web.ini file has four configuration sections - web, syscmds, node-overrides, and voter-titles.

    "},{"location":"allmon3/config/#web","title":"[web]","text":"

    The web section has the basic customizations for the Allmon3 site. Each item in this section is documented in the file.

    "},{"location":"allmon3/config/#syscmds","title":"[syscmds]","text":"

    The syscmds section defines the templates in the \"system commands\" menu. This stanza lists commands that are templates for the systems command modal dialog. The format is:

    [syscmds]\n  command text = Command Label\n  command text = Command Label\n  command text = Command Label\n  ...\n
    In any command text the @ will be replaced with the node the command modal was selected from. For example: 
        rpt status @ = Show Node Status\n
    wll result in the command 'rpt status 1999' assuming this command was selected from Node #1999.

    It is possible to create optional stanzas named in the format [syscmds-NODE] (e.g. [syscmds-1999]) and have those commands templates appear only for the given NODE listed. Example:

    [syscmds-1999]\nrpt cmd @ cop 999 xxx = Execute function 999\n
    "},{"location":"allmon3/config/#node-overrides","title":"[node-overrides]","text":"

    The node-overrides section can be used to override information from the ASL database. Using this section, the AllStarLink DB labels for a node may be replaces with custom text. The format is:

    [node-overrides]\n    NODE1 = TEXT HERE\n    NODE2 = TEXT HERE 2\n
    This section must exist even if it's empty. For example:

    [node-overrides]\n    1999 = My special private node\n    2010 = Wild Party Node\n
    "},{"location":"allmon3/config/#voter-titles","title":"[voter-titles]","text":"

    The voter-titles section is used to set display names for voters. This functions identically to [node-overrides]. For any voter not named here, voters will have an auto-generated name of \"Voter NODE\".

    "},{"location":"allmon3/config/#menuini","title":"menu.ini","text":"

    If this file is present in the api/ subdirectory of Allmon3's web interface and named menu.ini, this file will override the default behavior of the web interface simply listing the configured nodes in the left navbar. If this file is not present, all nodes in allmon3.ini/allmon3.ini.php will display singly in configured order.

    NOTE: Not listing a node in the menu WILL NOT cause the node to go un-polled if it is already configured in allmon3.ini. This allows the system administrator to have \"hidden\" nodes. If you want to completely stop polling an Asterisk/ASL node, the node must be removed from allmon3.ini.

    The format for this file is as follows:

    [ TITLE ]                         :: The stanza header is the label\n                                  :: for this menu item\ntype  = ( menu | single )         :: If type = menu, the item is interpreted as a dropdown\n                                  :: menu with each LABEL item displaying. If the\n                                  :: type = single, then only a the first LABEL = TARGET\n                                  :: will be displayed as a non-dropdown. In the case of\n                                  :: type = single, the [ TITLE ] is ignored completely\n\nLABEL = TARGET                    :: Each [ TITLE ] displays one or more LABELS\nLABEL = TARGET                    :: with a link to TARGET. When TARGET is all numbers\nLABEL = TARGET                    :: then TARGET is assumed to be a ASL node and\n                                  :: the link will filter down to the TARGET node\n                                  :: specified. Any other pattern will be interpreted\n                                  :: as a full or partial URL/URI.\n                                  ::\n                                  :: Of special node, creating a menu item of\n                                  :: multiple nodes to display, TARGET and simply be\n                                  :: #NODE,NODE,NODE - e.g. #1999,1998,1997\n
    Note that the order of the nodes within a stanza is irrelevant. They will be displayed as sorted alphabetically ascending according to UTF-8 (e.g. 0-9, A-Z, +).

    An example custom menu may look like: 

    [ W8WKY ]\ntype = menu\nW8WKY = 43211\n48496 = 48496\n45839 = 45839\n\n[ N8XPK ]\ntype = menu\nN8XPK = 42993\n43118 = 43118\n47987 = 47987\n\n[ Test ]\ntype = single\nAllStarLink = https://www.allstarlink.org\n

    "},{"location":"allmon3/config/#using-nginx-instead-of-apache","title":"Using Nginx instead of Apache","text":"

    Nginx can be used instead of Apache. Instead of using the apache2 package, install nginx using the above directions. After configuring nginx, edit /etc/nginx/sites-available/default (or your preferred site configuration) and add an include directive within the appropriate server { } configuration block. For example:

    server {\n    listen 80 default_server;\n\n    [... other stuff ...]\n\n    include /etc/allmon3/nginx.conf;\n\n    [... other stuff ...]\n}\n
    "},{"location":"allmon3/install/","title":"Installation","text":""},{"location":"allmon3/install/#supported-platforms","title":"Supported Platforms","text":"

    Allmon3 is only supported on:*

    "},{"location":"allmon3/install/#asl3-packages-install","title":"ASL3 Packages Install","text":""},{"location":"allmon3/install/#debian-12","title":"Debian 12","text":""},{"location":"allmon3/install/#debian-11","title":"Debian 11","text":""},{"location":"allmon3/install/#debian-10","title":"Debian 10","text":"

    Note: Allmon v1.2.1 will be the last version for Debian 10. Please upgrade your AllStarLink system to AllStarLink v3 on Debian 12.

    "},{"location":"allmon3/install/#install-from-source","title":"Install from Source","text":"

    Installation from source is not supported in the general use case. However, make install should yield a working system.

    "},{"location":"allmon3/usermgmt/","title":"User Management","text":"

    Allmon3 supports a robust multi-user configuration that permits per-node granular access as desired and that abstracts Allmon3 from the Asterisk AMI Manager configuration.

    Usernames and passwords are stored in /etc/allmon3/users.

    The default-configured username and password combination isallmon3 / password.

    You must change this before exposing a host to the Internet

    "},{"location":"allmon3/usermgmt/#user-database","title":"User Database","text":"

    Allmon3's user database is managed by allmon3-passwd. Adding a new user or editing an existing user is the same command. If the user does not exist, it will be added. If the user does exist, the password will be updated. To add or edit a user's password: 

    $ allmon3-passwd allmon3\nEnter the password for allmon3: password\nConfirm the password for allmon3: password\n
    That's all there is to it. The /etc/allmon3/users file is readable to see that the Argon2 hash changed for the user.

    Deleting a user is simply adding the --delete flag to the command:

    $ allmon3-passwd --delete allmon3\n

    After changing a user password, the allmon3 damon must be reloaded with systemctl reload allmon3.

    "},{"location":"allmon3/usermgmt/#per-node-restrictions-for-users","title":"Per-Node Restrictions for Users","text":"

    Allmon3 implements a lightweight access control system to restrict commands from certain users to certain nodes. Restrictions are configured in /etc/allmon3/user-restrictions. Given that the average use case is all users have similar access, the access control is implemented in a named-restrictions model for least configuration complexity.

    The logic is as follows when checking the restricted access list:

    1. If the user is not listed in user-restrictions at all than the user is permitted commands on all configured nodes.

    2. If the user is listed in user-restrictions and is listed as restricted to the node being commanded, the user is permitted to issue the command.

    3. If the user is listed in user-restrictions but the node is not listed for that user, the user is prohibited from issuing the command.

    The format of the user-restrictions file is:

    user1 | NODE[,NODE,NODE...]\nuser2 | NODE\n
    Lines beginning with # are comments.

    "},{"location":"developers/aptly/","title":"AllStarLink Apt Repos","text":"

    AllStarLink's primary installation method is through the use of Debian packages and the Apt deployment system.

    "},{"location":"developers/aptly/#repository-structure","title":"Repository Structure","text":"

    The repository is located at https://repo.allstarlink.org/public and is managed using Aptly. The Aptly structure is a multi-local-repo system that is coalesced into a single public repository with multiple components per distribution.

    There are three main local repositories:

    Each repository contains one or more distributions. Currently the only distribution supported is bookworm for Debian 12.

    Each repository has a default component name that, when published together, form the basis of the labeling scheme of promotion of packages from devel -> beta -> prod. The component mapping is:

    All local repositories are merged and published into a single public-facing repository. This is usable by one of the following .list configurations in/etc/apt/sources.list.d:

    # Primary AllStarLink Repo for Production Packages\ndeb [signed-by=/etc/apt/keyrings/allstarlink.gpg] https://repo.allstarlink.org/public bookworm main\n\n# Include Beta Component for Testing\ndeb [signed-by=/etc/apt/keyrings/allstarlink.gpg]  https://repo.allstarlink.org/public bookworm main beta\n\n# Include Beta and Devel component for Development\ndeb [signed-by=/etc/apt/keyrings/allstarlink.gpg]  https://repo.allstarlink.org/public bookworm main beta devel\n

    All packages in the repository are signed by GPG and the key is provided with repository installation .deb file.

    "},{"location":"developers/aptly/#aptly-github-runner-integrations","title":"Aptly + GitHub Runner Integrations","text":"

    ASL3-related GitHub repositories are integrated with the Aptly repository and running the \"Make and Publish Pkgs\" Action from each project will result in .deb files being compiled and stored in Aptly. Currently it's possible to build files directly into the devel and beta components but not prod. Moving packages into prod currently requires administrator intervention. This will be changed at some future date.

    "},{"location":"developers/aptly/#aptly-cli","title":"Aptly CLI","text":"

    It is possible to run Aptly commands from the shell on repo.allstarlink.org in addition to using the web-based API.

    "},{"location":"developers/aptly/#setup","title":"Setup","text":"

    All commands to Aptly must be run as the aptly user. To do this, each account on the repository server should be member of the sudo and aptly UNIX groups.

    Before issuing any Aptly commands run the following: 

    alias aptly=\"sudo -u aptly -- aptly\"\n
    to ensure that all commands are run as the aptly user.

    "},{"location":"developers/aptly/#adding-a-file-to-a-local-repo","title":"Adding a File to a Local Repo","text":"

    Adding a file to the local repo is straight-forward: 

    aptly repo add REPO FILE-or-DIR\n

    For example, to add the latest Allmon3 to the production repository: 

    $ aptly repo add asl3-prod allmon3_1.2.1-2_all.deb\nLoading packages...\n[+] allmon3_1.2.1-2_all added\n

    "},{"location":"developers/aptly/#publishing-repositories","title":"Publishing Repositories","text":"

    Coalescing and publishing the public repository with all components is: 

    aptly publish update bookworm\n
    When other distributions are supported, replace \"bookworm\" with the appropriate distribution. Upon execution, a prompt will appear for the GPG signing key for the repository. Provide it.

    And example, successful publication looks like: 

    $ aptly publish update bookworm\nLoading packages...\nGenerating metadata files and linking package files...\nFinalizing metadata files...\nSigning file 'Release' with gpg, please enter your passphrase when prompted:\nClearsigning file 'Release' with gpg, please enter your passphrase when prompted:\nCleaning up prefix \".\" components beta, devel, main...\n\nPublish for local repo ./bookworm [all, amd64, arm64, armhf] publishes {beta: [asl3-beta]}, {devel: [asl3-devel]}, {main: [asl3-prod]} has been successfully updated.\n

    "},{"location":"developers/aptly/#searching-for-packages","title":"Searching for Packages","text":"

    Listing and searching for packages inside of a local repo is done with the aptly repo search command and uses the (Aptly Package Query Language)[https://www.aptly.info/doc/feature/query/] to filter/select packages.

    A simple aptly repo search REPO will show all packages. For example: 

    $ aptly repo search asl3-prod\nallmon3_1.2.1-2_all\nasl-apt-repos_1.3-1.deb12_all\nasl3_3.0.0-1.deb_all\n[...]\n

    "},{"location":"developers/aptly/#search-by-name","title":"Search by name","text":"
    $ aptly repo search asl3-prod 'Name (asl3)'\nasl3_3.0.0-1.deb_all\n
    "},{"location":"developers/aptly/#search-by-name-glob-version","title":"Search by name glob + version","text":"
    $ aptly repo search asl3-prod 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12)'\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\n
    "},{"location":"developers/aptly/#search-by-name-glob-version-range","title":"Search by name glob + version range","text":"
    $ aptly repo search asl3-prod 'Name (~ asl3-asterisk*), Version (>= 1.0)'\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\n
    "},{"location":"developers/aptly/#promoting-packages-to-prod","title":"Promoting Packages to Prod","text":"

    Using the hypothetical example of promoting asl3-asterisk packages from beta to production.

    (1) Test package selection using aptly repo search 

    $ $ aptly repo search asl3-beta 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12)'\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64\n

    (2) Move the packages into the production repository using aptly repo move 

    $ aptly repo move asl3-beta asl3-prod 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12)'\nLoading packages...\n[o] asl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all moved\n[o] asl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all moved\n[o] asl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all moved\n[o] asl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n

    (3) Update the repository publication 

    $ aptly publish update bookworm\nLoading packages...\nGenerating metadata files and linking package files...\nFinalizing metadata files...\nSigning file 'Release' with gpg, please enter your passphrase when prompted:\nClearsigning file 'Release' with gpg, please enter your passphrase when prompted:\nCleaning up prefix \".\" components beta, devel, main...\n\nPublish for local repo ./bookworm [all, amd64, arm64, armhf] publishes {beta: [asl3-beta]}, {devel: [asl3-devel]}, {main: [asl3-prod]} has been successfully updated.\n

    "},{"location":"developers/aptly/#note-on-all-architecture-packages","title":"Note on 'all' Architecture Packages","text":"

    The architecture all denotes that the package contains no binary compiled code and runs on any architecture. This is meaningful when handling asl3-asterisk packages for multiple architectures. Within the same version and release only one copy of the all packages needs to and CAN exist in the repository. The GitHub Runner builder will delete the all packages for the arm64 build since the arm64 and amd64 builds should occur simultaneously.

    To delete an extra set of all packages that may have ended up in the repository: 

    $ aptly repo search asl3-beta 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12), Architecture (all)'\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\n\n$ aptly repo remove asl3-beta 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12), Architecture (all)'\nLoading packages...\n[-] asl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all removed\n[-] asl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all removed\n[-] asl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all removed\n

    "},{"location":"developers/package-builds/","title":"Building ASL3-Asterisk","text":""},{"location":"developers/package-builds/#building-deb-files","title":"Building .deb Files","text":"

    These directions outline how to create the .deb files via GitHub using the Action workflows and the ASL custom runner AMIs in AWS.

    Kicking off the build processes requires knowledge of all of the topics below before starting it.

    "},{"location":"developers/package-builds/#package-version-format","title":"Package version format","text":"

    All packages created by this repo have the following version format based on deb-version(7) manpage explanation.

    \n ${EPOCH}:${ASTERISK_VERSION}+asl3-${RPT_VERSION}-${PACKAGE_VERSION}\n\n|----+---|--------+--------------------------X-----+--------X\n     |            |                                |\n     |            +-- \"upstream-version\"           |\n     |                                             |\n     +-- \"epoch\"               \"debian-revision\" --+\n

    The values are defined as follows:

    A file generated from this repo using the versioning format above will be named, for example asl3-asterisk-20.8.1+asl3-3.0.0-1. Note that the epoch does not appear in the filename by debian convention.

    "},{"location":"developers/package-builds/#determining-asterisk-version","title":"Determining Asterisk Version","text":"

    Currently, this is structured to be built against Asterisk 20 Long Term Servicing (LTS). Do not use any version of Asterisk that is not the latest version of Asterisk 20 LTS. The version is listed at Download Asterisk. The version of Asterisk entered in the Actions launcher will cause the proper version of Asterisk 20 LTS to be downloaded and folded into the builder.

    "},{"location":"developers/package-builds/#locking-in-and-determining-the-app_rpt-version","title":"Locking In and Determining the app_rpt Version","text":"

    When a build of .deb files is initiated, the app_rpt repo should must be tagged prior to initiating the build process. The tag version should be in the format MAJOR.MINOR.PATCH as found in app_rpt.h.

    "},{"location":"developers/package-builds/#determining-the-package-version","title":"Determining the Package Version","text":"

    This should be a monotonically increasing integer starting with 1 reflecting a change in Asterisk Version + app_rpt. For example, if building Asterisk 20.8.1 with app_rpt v3.0.0, then the first build of a package should be 1, the second 2, and so on.

    However, if from the above example, app_rpt is now v3.1.0 or Asterisk becomes 20.9.0 then the package version should reset back to 1 and start over.

    "},{"location":"developers/package-builds/#create-a-github-release","title":"Create a GitHub Release","text":"

    Create a Release to store the .deb files in. The name of the release should follow the format:

    ASTERISK_VERSION_asl3-RPT_VERSION-PACKAGE_VERSION\n

    For the above example of Asterisk \"20.7.0\", app_rpt \"1.2\", and package version \"1\" the release tag and name would be: 

    20.7.0_asl3-1.2-1\n
    Note: GitHub acts oddly about + signs in the tags and release names so it's converted to an underscore.

    "},{"location":"developers/package-builds/#executing-the-github-action-to-build-packages","title":"Executing the GitHub Action to Build Packages","text":"

    The following steps build the Debian packages:

    Note: as noted above, you will choose the platform architecture when starting a workflow. All workflows will create the architecture specific packages. The \"amd64\" workflow also creates the architecture independent (\"all\") packages. What does this mean? If you only want \"amd64\" packages then you would only need to run the single workflow. But, if you want packages for an alternate architecture (e.g. \"arm64\") then will need to run a workflow for that architecture AND one for the \"amd64\" architecture.

    "},{"location":"developers/source-install/","title":"Source-Based Installation","text":"

    Source install is for developers or users who need to install AllStarLink 3 on unsupported hardware or operating systems. Installing ASL from source code is primarily for developers. Doing so will require you to download, compile, and install multiple projects. You will also need to be very comfortable using various development tools and the Linux CLI.

    The following instructions are for building ONLY Asterisk with ASL's app_rpt.

    "},{"location":"developers/source-install/#install-phreaknetsh-script","title":"Install phreaknet.sh script","text":"

    The phreaknet script compiles and patches Asterisk and DAHDI. You should install the script with the following commands :

    cd /usr/src\nwget https://docs.phreaknet.org/script/phreaknet.sh\nchmod +x phreaknet.sh\n./phreaknet.sh make\n

    Once installed, you can keep the script updated with :

    phreaknet update\n
    "},{"location":"developers/source-install/#install-asterisk-20-lts","title":"Install Asterisk 20 LTS","text":"

    Use -t or -b for developer mode. Both are optional.

    phreaknet install -d -b -v 20\n

    Asterisk should be running at this point but not app_rpt. Check the install with asterisk -r.

    "},{"location":"developers/source-install/#clone-asl3-repo","title":"Clone ASL3 repo","text":"
    cd /usr/src\ngit clone https://github.com/AllStarLink/app_rpt.git\n
    "},{"location":"developers/source-install/#install-asl3","title":"Install ASL3","text":"

    This script does a git pull of app_rpt and compiles the branch you are on.

    cd app_rpt\n./rpt_install.sh\n
    "},{"location":"developers/source-install/#install-asl3-configs","title":"Install ASL3 configs","text":"

    This adds ASL3 configs to the full set of Asterisk configuration files. ASL3 modules.conf limits what actually runs.

    cp /usr/src/app_rpt/configs/rpt/* /etc/asterisk\n

    After restarting Asterisk (or rebooting the system) you should now have a complete ASL3 install. You can use the following command as a quick check :

    asterisk -rx \"rpt localnodes\"\n
    You should see node 1999.

    "},{"location":"mans/asl-find-sound/","title":"asl-find-sound","text":""},{"location":"mans/asl-find-sound/#name","title":"NAME","text":"

    asl-find-sound - Display app_rpt-compatible sound devices

    "},{"location":"mans/asl-find-sound/#synopsis","title":"SYNOPSIS","text":"

    usage: asl-find-sound

    "},{"location":"mans/asl-find-sound/#description","title":"DESCRIPTION","text":"

    asl-find-sound - Display app_rpt-compatible sound devices

    "},{"location":"mans/asl-find-sound/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-find-sound/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-node-lookup/","title":"asl-node-lookup","text":""},{"location":"mans/asl-node-lookup/#name","title":"NAME","text":"

    asl-node-lookup - Display AllStarLink node information

    "},{"location":"mans/asl-node-lookup/#synopsis","title":"SYNOPSIS","text":"

    asl-node-lookup [ --help ] [ --verbose ] [ --ns <name-server> ] <node#>

    Required arguments :

    <node#> : ASL node number to query

    Optional arguments :

    --help : show help

    --verbose : report additional information including the DNS \"SOA\" and \"NS\" records

    --ns : issue DNS queries to the specified name server

    Note: this command will also report Asterisk/rpt node lookup results if exec'd as \"root\" (or \"asterisk\").

    "},{"location":"mans/asl-node-lookup/#description","title":"DESCRIPTION","text":"

    asl-node-lookup - Display AllStarLink node information

    "},{"location":"mans/asl-node-lookup/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-node-lookup/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Allan Nathanson and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-play-arn/","title":"asl-play-arn","text":""},{"location":"mans/asl-play-arn/#name","title":"NAME","text":"

    asl-play-arn - Play Amateur Radio Newsline

    "},{"location":"mans/asl-play-arn/#synopsis","title":"SYNOPSIS","text":"

    usage: `asl-play-arn [-h] --node [ --when WHEN ] [ --debug ]

    optional arguments:

    "},{"location":"mans/asl-play-arn/#description","title":"DESCRIPTION","text":"

    Basic use is either immediately from the command line:

    allstar-play-arn --node 1999\n

    Depending on the processing speed of the device and Internet connectivity, the start of playback may take a significant time. If the desire is for precision on the start time, use the --when command and execute allstar-play-arn a few minutes before the desired start time.

    The script is silent except on errors like all good Unix utilities. Some useful troubleshooting may be done with the --debug option if an error is not revealing.

    "},{"location":"mans/asl-play-arn/#sceheduling-the-playback","title":"Sceheduling the Playback","text":"

    The best way to schedule the playback is as follows as the root user.

    1. cp /usr/share/asl3/asl-play-arn.* /etc/systemd/system

    2. Edit the OnCalendar= entry of /etc/systemd/system/asl-play-arn.timer for the timing of the playback desired.

    3. Edit the value for \"NODE\" in /etc/systemd/system/asl-play-arn.service for the node to execute the playback upon.

    4. Enable the timer unit: 

      systemctl daemon-reload\nsystemctl enable asl-play-arn.timer\n

    "},{"location":"mans/asl-play-arn/#asteriskapp_rpt-configuration","title":"Asterisk/app_rpt Configuration","text":"

    Usually the timeout timer in app_apt will be too short to accomodate the playing of the full news file. The following commands can be added to /etc/asterisk/rpt.conf to enable and disable the TOT:

    907=cop,7    ; Time out timer enable\n908=cop,8    ; Time out timer disable\n

    These appear as commented-out options in the stock rpt.conf

    "},{"location":"mans/asl-play-arn/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-play-arn/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-say/","title":"asl-say","text":""},{"location":"mans/asl-say/#name","title":"NAME","text":"

    asl-say - Cause Asterisk to speak

    "},{"location":"mans/asl-say/#synopsis","title":"SYNOPSIS","text":"

    usage: asl-say -n NODE -w ( time | time24 | ip4 | ip6 )

    "},{"location":"mans/asl-say/#description","title":"DESCRIPTION","text":"

    asl-say will speak the one of the following things on the node specified with -n as directed by -w

    time - The current time

    time24 - The current time in 24-hour format

    ip4 - The first IPv4 address of the system

    ip6 - The first global-scope IPv6 of the system

    "},{"location":"mans/asl-say/#example","title":"EXAMPLE","text":"

    asl-day -n 1999 -w time will speak the current time on node 1999.

    "},{"location":"mans/asl-say/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-say/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-setup-dkms-mok/","title":"asl-setup-dkms-mok","text":""},{"location":"mans/asl-setup-dkms-mok/#name","title":"NAME","text":"

    asl-setup-dkms-mok - Configure the Machine Owner Key (MOK) for signing kernel packages, notably dahdi* for ASL.

    "},{"location":"mans/asl-setup-dkms-mok/#synopsis","title":"SYNOPSIS","text":"

    usage: asl-setup-dkms-mok

    "},{"location":"mans/asl-setup-dkms-mok/#description","title":"DESCRIPTION","text":"

    This script is a walkthrough script for generating a UEFI Machine Owner Key (MOK). A MOK is needed on systems with UEFI SecureBoot to properly authenticate the DAHDI-related kernel modules which are rebuilt on kernel upgrades with DKMS. See the AllStarLink manual for more information or https://wiki.debian.org/SecureBoot.

    "},{"location":"mans/asl-setup-dkms-mok/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-setup-dkms-mok/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"pi/cockpit-basics/","title":"Cockpit Basics","text":"

    The Cockpit web console is a web-based graphical interface By using it, you can perform system administration tasks, such as inspecting and controlling services, managing storage, configuring networks, analyzing network issues, and inspecting logs. Of course, it's also very handy for administering Asterisk on your AllStarLink Pi Appliance!

    "},{"location":"pi/cockpit-basics/#what-is-the-cockpit-console","title":"What is the Cockpit Console","text":"

    What is the Cockpit Console

    The Cockpit Console is a web-based interface designed for managing and monitoring your AllStarLink Pi Appliance.

    The Cockpit Console enables you to perform a wide range of administration tasks, including:

    Managing AllStarLink, Asterisk, and its related applications\nManaging services\nManaging user accounts\nManaging and monitoring system services\nConfiguring network interfaces\nReviewing system logs\nUpdating software\n

    The Cockpit Console uses the same system concepts as you would use in a terminal, and actions performed in a terminal are immediately reflected in the Cockpit Console. That is to say, you can use Cockpit and an SSH connection interchangeably at your desire.

    You can monitor the logs of application. In addition, you can change the settings directly in the web console or through the terminal.

    Throughout this documentation a host named node63001.local is used as the example host. Substitute that name with your own node's name as it was set during the imaging process.

    "},{"location":"pi/cockpit-basics/#logging-in-to-the-web-console","title":"Logging In to the Web Console","text":"

    1. In your web browser, enter the following address to access the web console: https://node63001.local:9090

    2. The browser may display a warning. Accept the security exception to proceed with the login.

    3. Enter the username and password set during the imaging process.

    4. Click Log In.

    "},{"location":"pi/cockpit-basics/#administrative-access","title":"Administrative Access","text":"

    The Web Console has a concept of Administrative access. While it does remember the access level across logins, to access certain functions you may need to enable Administrative access from Limited access. Do do that:

    1. Click on the bar marked Limited Access to enable admin-level access.

      You may be prompted to tnter the password again. Do so and then click Authenticate

    2. Click Close on the box reporting administrative access has been granted.

    "},{"location":"pi/cockpit-basics/#shutting-down-the-appliance","title":"Shutting Down the Appliance","text":"

    You can use the web console to shut down the system.

    1. Log into the Web Console.

    2. (optional) Click on the bar marked Limited Access to enable admin-level access.

    3. (optional) Click Close on the box reporting administrative access has been granted.

    4. Click on the downward arrow next to Reboot and then select Shutdown

    5. Change the Delay dropdown to No delay and click Shut down

    6. A Disconnected message will appear. The node will shutdown is about 15-30 seconds, after which power may be removed.

    "},{"location":"pi/cockpit-basics/#rebooting-the-appliance","title":"Rebooting the Appliance","text":"

    1. Log into the Web Console.

    2. (optional) Click on the bar marked Limited Access to enable admin-level access.

    3. (optional) Click Close on the box reporting administrative access has been granted.

    4. Click on Reboot

    5. Change the Delay dropdown to No delay and click Reboot

    6. A Disconnected message will appear. The node will reboot and be available again in about 15-30 seconds. After that, one can click Reconnect.

    "},{"location":"pi/cockpit-basics/#configuring-time-on-the-appliance","title":"Configuring Time on the Appliance","text":"

    In general, the appliance should be left in the default setting of using the NTP service to automatically set the time. However you may want to change the Timezone. To change the time:

    1. Log into the Web Console.

    2. (optional) Click on the bar marked Limited Access to enable admin-level access.

    3. (optional) Click Close on the box reporting administrative access has been granted.

    4. Click on Overview and scroll down to Configuration

    5. Click on highlighted time next to System time

    6. Change the Time zone drop-down to the desired zone.

    7. Click Change.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-console/","title":"Terminal / Console","text":"

    The Cockpit Web Console provides a feature-complete terminal interface in the web browser without requiring setup and configuration of an SSH client. This allows administration of the node from anywhere!

    Use of the web console Terminal and an SSH client is fully interactive and interchangeable.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Terminal in the web console menu on the left.

    3. You will be connected to a terminal of the user that you logged in with.

    4. To execute commands as root, either prefix the command with sudo:

    5. One can enter a root session completely by issueing the command sudo -s:

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-firewall/","title":"Managing the Firewall","text":"

    A firewall is a way to protect nodes from any unwanted traffic. It enables users to control incoming network traffic on host machines by defining a set of firewall rules. These rules are used to sort the incoming traffic and either block it or allow it through. In the AllStarLink Pi APpliance, the firewalld service with the nftables back end functions as the firewall. Through the web console, you can configure firewalld.

    "},{"location":"pi/cockpit-firewall/#adding-an-inbound-port","title":"Adding an Inbound Port","text":"

    The firewall is enabled by default with a reasonable set of default rules. The most common reasons to edit the port list would be to change the IAX2 port to be outside the range of 4560-4580 (default is 4569) or to change the voter port (default 1667 in ASL3, might need 667 for legacy reasons).

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. Click Edit rules and zones in the Firewall panel.

    4. Click Custom ports

    5. Enter the TCP or UDP port or ports to be permitted in the TCP or UDP fields respectively. Multiple ports can be separated with commands and a range of ports can be specified with a hyphen. It is highly suggested to include a Description for the port. For example, if adding an alternative IAX2 port on 14569:

    6. Click Add ports

    7. The change is immediate.

    "},{"location":"pi/cockpit-firewall/#removing-an-inbound-port","title":"Removing an Inbound Port","text":"

    The firewall is enabled by default with a reasonable set of default rules. The most common reasons to edit the port list would be to change the IAX2 port to be outside the range of 4560-4580 (default is 4569) or to change the voter port (default 1667 in ASL3, might need 667 for legacy reasons).

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. Click Edit rules and zones in the Firewall panel.

    4. Next to the service to delete, click on the three vertical dots and select Delete. The change is immediate.

    "},{"location":"pi/cockpit-firewall/#enabling-and-disabling","title":"Enabling and Disabling","text":"

    The firewall is enabled by default with a reasonable set of default rules. Unless you are familiar with networking, DO NOT disable the firewall or make changes to it.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. To enable the firewall, slide the toggle to the right. This is the default setting and should not be changes unless you're 100% sure of what you are doing.

    4. To disable the firewall, slide the toggle to the left. Donot disable the firewall unless you're 100% sure of what you are doing.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-get-started/","title":"Pi Appliance","text":"

    AllStarLink 3 features a modern appliance of AllStarLink while retaining the ubiquitous Raspberry Pi experience that's well-documented and familiar to the amateur radio community worldwide.

    "},{"location":"pi/cockpit-get-started/#cockpit-web-admin-interface","title":"Cockpit Web Admin Interface","text":"

    The AllStarLink Pi Appliance includes the user-friendly Cockpit system for ease of administration. Cockpit is access on port 9090 of the appliance. In these examples the hostname node63001 is used and should be replaced with the name you set during setup. If the name node63001 was used at setup, then pointing a browser to https://node63001.local:9090/ will bring up the Cockpit interface. Note: The hostname only works when your node and your PC are on the same LAN. Otherwise use the IP or dns name you assign. Login is the username and password that was setup during the Pi imaging process.

    The first connection will report a message that \"Your connection isn't private\". For the Pi appliance, this is acceptable. Click on Advanced and then Continue to node63001.local (unsafe). This only must be done the first time.

    Enter the username and password configured during the imaging process above. Click Log in

    Cockpit is fairly intuitive to use, but here's are a few pointers to help get started:

    "},{"location":"pi/cockpit-network/","title":"Network Configuration","text":"

    The Networking system tool in the web console permits viewing the status of the network as well as making network configuration changes to the appliance.

    "},{"location":"pi/cockpit-network/#observing-network-status-and-configuration","title":"Observing Network Status and Configuration","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. At the top of the screen will be the network data statistics for transmitted and receiving network traffic. This information can be helpful if there are questions about network bandwidth utilization:

    4. The next panel down shows the current configuration of the present network adapters:

    5. Finally, at the bottom is the logs about the current state of the network:

    "},{"location":"pi/cockpit-network/#changing-adapter-setting","title":"Changing Adapter Setting","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. Scroll down to the Interfaces section

    4. Click on the Name of the interface

    5. Make desired changes

    6. Changes take place immediately. Be cautious when changing IP addressing because you may accidentally remove your ability to access to the console.

    "},{"location":"pi/cockpit-network/#changing-wifi-settings","title":"Changing WiFi Settings","text":"

    Unfortunately, Cockpit does not have an interface for managing WiFi networks with in it. Currently WiFi network changes must be made from the Web Console Terminal screen.

    "},{"location":"pi/cockpit-network/#listing-existing-wifi-networks","title":"Listing Existing WiFi Networks","text":"

    List the existing WiFi network(s) with nmcli device wifi list:

    "},{"location":"pi/cockpit-network/#add-a-wifi-network","title":"Add a WiFi Network","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Terminal in the web console menu on the left.

    3. At the prompt enter the command:

      nmcli device wifi connect MYNETWORK password MYPASSWORD\n

      where MYNETWORK is the name (\"SSID\") and MYPASSWORD is the password (\"Key\"). To join the network \"RadioSite\" with the password \"RF1234\":

      nmcli device wifi connect RadioSite password RF1234\n

      If there is a space in the name or password, you must use quotes around the element with the space in it.

    4. If the network is not currently active (i.e. adding a network where the appliance will be in the future) or the SSID is hidden, adding is slightly more complicated. Again, substitute MYNETWORK and MYPASSWORD with as appropriate.

      nmcli conn add type wifi con-name MYNETWORK ifname wlan0 ssid MYPASSWORD\nnmcli conn modify MYNETWORK wifi-sec.key-mgmt wpa-psk wifi-sec.psk MYPASSWORD\n

    Note: DO NOT use the raspi-config tool to attempt to set the Wifi network and password. This will break the appliance's networking.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-services/","title":"Services Management","text":"

    The Cockpit Web Console allows the management of all of the running services. You can activate or deactivate services, restart or reload them, or manage their automatic startup.

    "},{"location":"pi/cockpit-services/#start-stop-and-restart-services","title":"Start, Stop, and Restart Services","text":"

    This procedure stops or starts system services using the web console interface. You can filter the services by name or description and also by Enabled, Disabled, or Static automatic startup. The interface shows the current state of the service and its recent logs.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Services in the web console menu on the left. The default tab for Services is System Services. If you want to manage targets, sockets, timers, or paths, switch to the respective tab in the menu on top. To open service settings, click on a selected service from the list. You can tell which services are active or inactive by checking the State column.

    3. To start an inactive service, click the Start button.

    4. To deactivate an active service, click the Stop button

    5. To restart a service, click the Restart button

    "},{"location":"pi/cockpit-services/#seeing-status-and-recent-logs-of-a-service","title":"Seeing Status and Recent Logs of a Service","text":"

    This procedure stops or starts system services using the web console interface. You can filter the services by name or description and also by Enabled, Disabled, or Static automatic startup. The interface shows the current state of the service and its recent logs.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Services in the web console menu on the left. The default tab for Services is System Services. If you want to manage targets, sockets, timers, or paths, switch to the respective tab in the menu on top. To open service settings, click on a selected service from the list. You can tell which services are active or inactive by checking the State column.

    3. Scroll down to see Service logs. The botton View All Logs can be clicked to review an extensive output of the service's logs.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-updates/","title":"Software Updates","text":"

    The Software updates tool in the web console is how to install package updates to the AlLStarLink Pi appliance. This includes updating AllStarLink software and its associated items.

    "},{"location":"pi/cockpit-updates/#checking-for-and-installing-updates","title":"Checking For and Installing Updates","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Software Updates in the web console menu on the left.

    3. Click on the cycle icon in the upper right of the panel.

    4. Available updates will be displayed. Click on Install all updates

    5. The updates will start. Depending on the quantity of updates, the speed of the Pi hardware, and how long certain installs take, this may take quite awhile to copmlete. For example, if there is a Linux kernel update, then the Dahdi package will be recompiled for the new kernel which can take up to 10 minutes. Be patient while updates run. You can observe the updates by clicking on View update log.

    6. When the updates are complete, you will be prompted to reboot the system. In general, it is advised to reboot the system after updates, however that is not necessarily required for most updates. Kernel updates and Dahdi updates DO require a reboot.

    7. After the updates are completed, returning to the Software Updates screen should show that the software is up to date.

    "},{"location":"user-guide/","title":"ASL3 User Guide","text":"

    This is the AllStarLink version 3 User's Guide. No doubt you're you've noticed this is not a Wiki. The ASL Wiki still exists, largely to document ASL1 and ASL3, while this manual covers ASL3 specifics and is the future home for all ASL documentation.

    Did you see the \"Next\" and \"Previous\" menu items? Those will page to the next section of this document in the order of the bullets below. We think you'll find this an orderly way to learn about ASL3. Lets get started.

    "},{"location":"user-guide/asl-cmds-tools/","title":"ASL Commands and Tools","text":"

    The following commands and tools are the primary AllStarLink ecosystem. Each entry contains a description of the command, link to the documentation, and what package provides the command or tool.

    Command Description Documentation Package asterisk The main console for Asterisk and app_rpt Normally invoked with asterisk -r asl3-asterisk asl-menu The main text-based menu for configuring ASL Menu Use asl3 asl3-update-nodelist Primary engine to update the /var/lib/asterisk/rpt_extnodes database like for the node dictionary. Node Resolution asl3-update-nodelist asl3-update-astdb Universal engine to provide astdb.txt to various applications that require the old node name database. Use this rather than a special version for every third-party app (e.g. Supermon, Node Remote, etc.) Node Resolution asl3-update-nodelist allmon3-passwd Manage the user database and passwords for Allmon3 Allmon3 User Management allmon3 asl-find-sound Utility to help locate compatible sound devices manpage asl3 asl-node-lookup Utility to test node lookup when troubleshooting Asterisk/app_rpt connectivity issues manpage asl3 asl-play-arn Tool to play Amateur Radio Newsline manpage asl3 asl-repo-switch Utility to change package release streams for AllStarLink Beta Testing asl3 asl-say Tool for basic statements into Asterisk like the sime manpage asl3 asl-show-version Utility to report the package version of all related ASL3 software. Normally used in bug reports and support requests. N/A asl3 asl-tts Text-to-speech engine for AllStarLink Text to Speech asl3-tts sa818 CLI tool for configuration SA818/DRA818-based radio modules. Use `sa818-menu in most cases. SA818 Page asl3 sa818-menu Menu for configuring SA818/DRA818-based radio modules SA818 Page asl3"},{"location":"user-guide/betatest/","title":"Beta Testing","text":"

    Beta testing is the final phase of software testing before releasing a product to the general public. During this stage, a group of users representing the target audience validates the product. They explore its features, uncover bugs, and provide feedback.

    AllStarLink releases software packages to a beta channel on a periodic basis to address bug fixes and feature enhancements. The beta channel of packages is available to all users. However, please only select to run packages from the beta channel if you meet the following criteria:

    1. If something goes wrong, your can easily recover your node from backups or other saved configuration.

    2. You have generalized Linux skills such as understanding basic commands, ability to use the shell terminal of Linux, ability to copy/paste output data into bug reports, etc.

    3. You acknowledge that a beta release may have other bugs, incomplete fixes, or may in rare cases not work right at all for your configuration and that you can live with this.

    4. You may need to edit configuration files to fix issues or test things or to roll back to previous versions of software.

    If you cannot adhere to the above four conditions, you should not use the beta channel software.

    "},{"location":"user-guide/betatest/#enabling-beta-channel-software","title":"Enabling Beta Channel Software","text":"

    Enabling beta channel software is done with the asl-repo-switch command as root. Set the channel to \"beta\" using asl-repo-switch -l beta. For example:

    # asl-repo-switch -l beta\nSetting level beta... DONE\nRun 'apt update' to refresh the package repo cache.\n
    "},{"location":"user-guide/betatest/#disabling-beta-channel-software","title":"Disabling Beta Channel Software","text":"

    Disabling beta channel software is done with the asl-repo-switch command as root. Set the channel to \"main\" using asl-repo-switch -l main. For example:

    # asl-repo-switch -l main\nSetting level main... DONE\nRun 'apt update' to refresh the package repo cache.\n
    "},{"location":"user-guide/betatest/#installing-updates-using-beta-channel","title":"Installing Updates Using Beta Channel","text":"

    The most common case to use the beta channel for software packages is to test a proposed fix to a problem you are having. In those cases, the following commands should be run one at a time from a terminal/console as root (using sudo or sudo -s):

    sudo asl-repo-switch -l beta\nsudo apt update\nsudo apt upgrade -y\nsudo asl-repo-switch -l main\n

    This will do a one-time installation of the packages in the beta channel and then switch back to main.

    "},{"location":"user-guide/betatest/#resetting-installation-of-asl-back-to-production","title":"Resetting Installation of ASL Back to Production","text":"

    To reset your packages for ASL3 back to the production-release main versions:

    For Pi Appliances: 

    sudo asl-repo-switch -r\nsudo apt update\nsudo apt install -y --reinstall asl3 asl3-asterisk asl3-asterisk-config \\\n  asl3-asterisk-modules asl3-menu asl3-update-nodelist \\\n  asl3-pi-appliance allmon3\n

    For Debian 12 general installations: 

    sudo asl-repo-switch -r\nsudo apt update\nsudo apt install -y --reinstall asl3 asl3-asterisk asl3-asterisk-config \\\n  asl3-asterisk-modules asl3-menu asl3-update-nodelist\n

    For DAHDI Kernel packages after doing the above, as necessary: 

    sudo apt install -y --reinstall dahdi-linux dahdi-dkms\n

    Note that you may need to merge or fix configurations when rolling back from beta software to production. This is unusual but possible.

    "},{"location":"user-guide/install/","title":"Debian 12 Install","text":"

    These instructions are for installing ASL3 on general purpose operating systems manually. For installation on Raspberry Pis, consider using the AllStarLink 3 Pi Appliance

    Architecture Support

    Currently, the project does not support armv7l/armhf platforms because all known uses of AllStarLink is on hardware which supports the Bookworm arm64 distribution such as Raspberry Pi 3, 4, 5, and Zero 2 W. If you have a platform that can use armv7l/armhf 32-bit packages only please file an issue at ASL3 on GitHub.

    Asterisk/app_rpt on VMs/Cloud

    Asterisk/app_rpt works fine on virtual machines in a cloud environment (often terms \"VPS\"). However Asterisk/app_rpt requires certain kernel modules to function properly and these are provided by ASL3 in the dkms-dahdi package. There are, however, two situations where Asterisk/app_rpt will not work:

    1. When the VPS provider is using a customized Debian using the \"cloud\" kernel. If the output of uname -r refers to the kernel as something similar to \"6.1.0-12-cloud-x86_64\" then you must follow this procedure prior to installing ASLv3.

    2. When the virtual machine is using an emulated Linux kernel. Microsoft Windows Subsystem for Linux (WSL2) is a good example of this situation. The kernel is custom and not related to the operating system distribution. Asterisk/app_rpt cannot work in this type of setup.

    "},{"location":"user-guide/install/#system-requirements","title":"System Requirements","text":"

    The following are the system requirements for an ASL3 system:

    Minimum Required Recommended CPU/Platform 1 CPU, 64-bit, x86_64 (amd64) or ArmV8 (arm64) 2 - 4 CPUs depending on the number of hardware devices connected to the system Memory 512M 2 G Storage 8G (for OS + software) -

    UEFI / SecureBoot

    For x86_64/amd64 platforms, it is recommended to disable UEFI and SecureBoot if you do not need those features and your platform supports a \"legacy BIOS\" mode. While they are a good security feature, given that AllStarLink v3 requires building a kernel module, it adds likely-undesired complexity for most ASL users. If you need or want to use UEFI/SecureBoot see the advanced topic document.

    "},{"location":"user-guide/install/#os-install","title":"OS Install","text":"

    You\u2019re going to start off by installing a new Debian 12 OS on your PC computer or virtual machine. There are instructions all over the internet that detail how to install Debian 12. Briefly you:

    Note: When setting up users and passwords you may be prompted to set the password for the 'root' account. If you do not provide a password then the root account will be disabled and the system's initial user account (that will requested on the next screen) will be given the power to become root using the \"sudo\" command. If you do provide a password then the \"sudo\" command will not be installed on your system. Our recommendation is to NOT set the 'root' password and rely on using the \"sudo\" command only when needed.

    "},{"location":"user-guide/install/#allstarlink-package-repo-install","title":"AllStarLink Package Repo Install","text":"

    Once your Debian system is up and running, install the ASL package repositories:

    cd /tmp\nwget https://repo.allstarlink.org/public/asl-apt-repos.deb12_all.deb\nsudo dpkg -i asl-apt-repos.deb12_all.deb\nsudo apt update\n
    "},{"location":"user-guide/install/#asl3-packages-install","title":"ASL3 Packages Install","text":"

    Now the packages may be installed and updated directly from the AllStarLink package repository:

    sudo apt install asl3\n

    This will install the complete ASL3 system including all of the Asterisk app_rpt-enabled packages (asl3-asterisk, asl3-asterisk-config, asl3-asterisk-doc, asl3-asterisk-modules), the Dahdi kernel module (dahdi-dkms, dahdi-linux, dahdi-source), the needed development tools to keep the kernel module updated, and the asl3-menu.

    It's important to note that the new packaging format will allow for easy, automated updates of kernels through the standard apt upgrade process. There is no longer any reason to hold back kernel upgrades with the ASL3 packaging.

    The following packages are also available and may be installed separately:

    Allmon3 - The updated web interface to AllStarLink which includes strong support for mobile devices and screens of all sizes and shapes. See the Allmon3 page for details.

    ASL3 Nodelist Updater - An updated node service to maintain a local copy of the AllStarLink node database (rpt_extnodes). While, in general, the preferred method of node lookup is DNS, some installations with slow Internet or DNS servers that have very long cache timers may benefit from this method of node lookup instead. Install with sudo apt install asl3-update-nodelist.

    "},{"location":"user-guide/install/#node-configuration","title":"Node Configuration","text":"

    Next step is to configure the node settings. It is recommended to use the asl-menu command to manage the configuration for common use cases. See ASL3 Menu for details. YouTuber Freddie Mac has a nice ASL3 RPi installation and configuration video. See the part where the asl-menu is shown.

    "},{"location":"user-guide/install/#docker","title":"Docker","text":"

    Docker support will be coming in the near future.

    "},{"location":"user-guide/known-issues-diffs/","title":"Known Issues & Differences With Legacy Versions","text":"

    As discussed in What's New?, AllStarLink v3 is a gigantic leap forward in platform and software ecosystem. Many things are different in modern versions of Linux, Asterisk, and more. This page documents permanent differences with ASL3 relative to legacy versions as well as any known issues that must be fixed in ASL3.

    "},{"location":"user-guide/known-issues-diffs/#differences-with-legacy-versions","title":"Differences with Legacy Versions","text":""},{"location":"user-guide/known-issues-diffs/#serial-ports-available-by-default","title":"Serial Port(s) Available by Default","text":"

    On the ASL3 Pi Appliance, the system comes pre-configured for /dev/serial0 (formerly /dev/ttyAMA0) accessibility. That means that Bluetooth and the default serial console are disabled. Any directions requiring editing of config.txt or cmdline.txt are unnecessary with the ASL3 appliance.

    "},{"location":"user-guide/known-issues-diffs/#pi-dev-entry-changes","title":"Pi /dev Entry Changes","text":"

    As ASL3 is based on Debian 12, users with Raspberry Pi devices must note that the serial port on the Pi header is now /dev/serial0 rather than the historical /dev/ttyAMA0. If you are following directions for Pi serial port operations, such as programming an SA818/DRA818-based radio hat or a SHARI node, use /dev/serial0 in place of the /dev/ttyAMA0 reference.

    "},{"location":"user-guide/known-issues-diffs/#voterrtcm-default-port","title":"Voter/RTCM Default Port","text":"

    Modern installations of Asterisk runs as the unprivileged asterisk user rather than as root. Standard Linux convention prohibits non-root users to listen on a TCP port below 1024. The default port for Voters/RTCMs has been changed to 1667 when previously it was 667. If voter port changes are difficult for the environment, see Incompatible Changes in ASL3 for other potential workarounds.

    "},{"location":"user-guide/known-issues-diffs/#known-issues","title":"Known Issues","text":"

    The following issues are currently known to exist in AllStarLink 3 and, where possible what the workarounds are.

    "},{"location":"user-guide/known-issues-diffs/#dtmf-linking-dns-lookup-timeouts","title":"DTMF Linking + DNS Lookup Timeouts","text":"

    There are intermittent cases of linking problems in the following situation:

    1. When a linking command (e.g. *3) is sent in DTMF; AND
    2. Certain network configurations or situations do not resolve DNS \"fast enough\"

    This issue is fully described in app_rpt GitHub issue #392. In summary, if the DTMF entry, decoding, and DNS resolution for a link command collectively takes longer than 3 seconds, linking can fail in silent and indeterminate ways.

    If this problem is occurring consistently, changing the node lookup method to \"file\" works around the issue until updated code to resolve this problem can be developed and tested:

    1. Install the asl3-update-nodelist package (not needed for those running the ASL3 Pi Appliance):

      sudo apt install -y asl3-update-nodelist\n

    2. Edit /etc/asterisk/rpt.conf. Somewhere at or around line 10 (depending on file customizations) you will need to change the node_lookup_method to file instead of both. After your change the file shoud look like:

      [general]\nnode_lookup_method = file       ;method used to lookup nodes\n                                ;both = dns lookup first, followed by external file (default)\n                                ;dns = dns lookup only\n                                ;file = external file lookup only\n

      This file can be updated using asl-menu \"Expert Configuration Menu\" / \"Edit rpt.conf file\" or your favorite editor.

    3. Restart Asterisk using asl-menu \"Node Settings\" / \"Restart Asterisk\" or systemctl restart asterisk.

    Once the fix is completed and made available, it will be announced on the ASL Community and you should change node_lookup_method to both. Eventually the \"file\" method will be retired.

    "},{"location":"user-guide/known-issues-diffs/#resize2fs_once-error","title":"resize2fs_once \"Error\"","text":"

    There are intermittent cases of errors on the screen or in the system logs about a failure of a service named resize2fs_once.service after the final first boot upon installation. The error may report that it \"Failed to start\" or \"timed out\". If the / partition has been properly resized - which has been the case in every known occurrence of the error - then there is no action to take and the issue will not appear on subsequent reboots.

    A properly resized / should be a bit smaller than the full size of the SD card or USB drive used with the device. In Cockpit, look at the Storage tab:

    In this example, / is a 31G partition on a 32G SD card.

    "},{"location":"user-guide/known-issues-diffs/#shutdownrestart-issues-with-ezstreambroadcastify-enabled","title":"Shutdown/Restart Issues with EZStream/Broadcastify Enabled","text":"

    With the configuration for Broadcastify enabled, shutting down or restarting Asterisk is not quick or always reliable. When issuing systemctl stop or systemctl restart for the asterisk.service unit it will take upwards of 90 seconds to properly terminate and exit/respawn. Make sure any lame and ezstream processed are killed and not orphaned.

    # ps auxww | grep asterisk\nasterisk 1443572  0.0  0.0   2416  1024 ?        S    10:13   0:00 /bin/sh -c /usr/bin/lame --preset cbr 16 -r -m m -s 8 --bitwidth 16 - - | /usr/bin/ezstream -qvc /etc/ezstream.xml\nasterisk 1443573  0.1  0.1  11920  5120 ?        S    10:13   0:00 /usr/bin/lame --preset cbr 16 -r -m m -s 8 --bitwidth 16 - -\nasterisk 1443574  0.0  0.2  55232 10240 ?        S    10:13   0:00 /usr/bin/ezstream -qvc /etc/ezstream.xml\n

    If Asterisk is shutdown, the processes above must be stopped with a kill -9 PID command.

    "},{"location":"user-guide/menu/","title":"ASL3 Menu","text":"

    Once you have installed ASL3 on your system you will want to configure your node. The ASL3 menu will help you get started with many common configurations including nodes with USB audio interfaces and \"hub\" nodes. The ASL3 menu will also help you maintain your system.

    Your system may have been setup to automatically start the ASL3 menu as soon as you have logged in. If not, you can access the ASL3 menu by executing the following command from the Linux shell :

    asl-menu\n

    Note: if you are not logged in as the \"root\" user you will need to use

    sudo asl-menu\n

    When the ASL3 menu is running you will be presented with options to configure a node, access the Asterisk CLI, backup (or restore) your ASL settings, access the Linux CLI, perform some basic diagnostics, and perform some common system actions.

    "},{"location":"user-guide/menu/#node-settings","title":"Node settings","text":"

    For new installs you will want to select the ASL3 \"Node Settings\" menu option. This is where you will configure the basic settings needed to get your new node on the air. You can return to this menu later to review and/or update your node settings.

    The \"Node Settings\" menu allows you update key node settings including :

    The \"Node Settings\" menu also allows you to add (or remove) additional nodes.

    Note: you can also use the node-setup command to access this menu.

    "},{"location":"user-guide/menu/#asterisk-cli","title":"Asterisk CLI","text":"

    The Asterisk CLI provides direct access to the heart of what makes an AllStar node. You might use the CLI for testing, troubleshooting, or for controlling your node. You should know that many of the commands require a more in-depth knowledge of Asterisk. The good news is that many of the commonly used commands can be executed from the \"Diagnostics Menu\" or with one of the web management/monitoring applications (e.g. Allmon3).

    "},{"location":"user-guide/menu/#backup-and-restore","title":"Backup and Restore","text":"

    The ASL3 \"Backup and Restore\" menu option provides a simple option to backup (and restore) your AllStar and Asterisk configuration. The backup archives are stored locally on your system and optionally in the cloud.

    We encourage you to \"backup\" your configuration, especially before and after you make changes.

    Note: you can also use the asl-backup-menu command to access this menu.

    "},{"location":"user-guide/menu/#diagnostics","title":"Diagnostics","text":"

    The ASL3 \"Diagnostics\" menu option allows you to perform some simple/rudimentary checks of your system and node. These include basic internet connectivity checks and reporting of your AllStar registrations. We also provide a way to stop, start, and restart some of the AllStar and Asterisk services that run on your system.

    Note: you should know that these diagnostics are not exhaustive. With a good connection to the internet and a valid configuration all \"should be\" OK. If the provided diagnostics don't flag an issue then you will need to dig deeper. Be aware of changes you make to your system and configuration. Be aware of software packages that you install. Be aware of \"upgrades\". If you need help, reach out to the AllStarLink Community.

    "},{"location":"user-guide/menu/#expert-configuration","title":"Expert Configuration","text":"

    The ASL3 \"Expert Configuration Menu\" option will allow you to make changes to some of the AllStar and Asterisk configuration files (in the \"/etc/asterisk\" directory). But, don't worry about that advanced stuff just yet.

    "},{"location":"user-guide/menu/#bash-shell","title":"Bash shell","text":"

    The \"bash\" shell provides access to the Linux command line interface, also known as the Linux CLI. This is for experts or if you have been provided with CLI instructions.

    "},{"location":"user-guide/passwords/","title":"Logins and Passwords","text":"

    Yes, there are many \"different\" logins and passwords associated with AllStarLink, the ASL3 software, and the related applications. A typical ASL3 server / node will directly or indirectly use the following login/password pairs :

    "},{"location":"user-guide/passwords/#allstarlink-website-wwwallstarlinkorg","title":"AllStarLink website (www.allstarlink.org)","text":"

    Th AllStarLink website is where you register for an AllStarLink account, add ASL servers, and request ASL nodes.

    "},{"location":"user-guide/passwords/#linux","title":"Linux","text":"

    Your desktop, laptop, Raspberry Pi, or Virtual Machine will be running the Linux operating system (e.g. Debian). To access the operating system on your OS you will need a login \"user\" and \"password\".

    You will use the Linux \"user\" and \"password\" when logging in to your system :

    You may have heard about the Linux \"root\" account. The root account is also called the super-user; it is a login that bypasses all security protection on your system. The root account should only be used to perform system administration, and only used for as short a time as possible. With ASL3, we are trying to limit our usage of the \"root\" account.

    Note: there is NO DEFAULT USER included in the ASL3 Raspberry Pi Appliance image.

    "},{"location":"user-guide/passwords/#asterisk-manager-interface-ami","title":"Asterisk Manager Interface (AMI)","text":"

    Your AllStarLink server system runs as process named \"asterisk\" that is the heart of your node(s). The Asterisk Manager Interface, AMI, is the standard management interface into your Asterisk server.

    The AMI configuration, stored in \"/etc/asterisk/manager.conf\", includes \"user\" categories and associated \"secret\" passwords. For example :

    [admin]\nsecret = my-AMI-password\n...\n

    This \"user\" (admin) and \"secret\" (my-AMI-password) would need to be specified in the configuration file for any applications that use AMI to interact with Asterisk.

    "},{"location":"user-guide/passwords/#web-applications-eg-allmon3","title":"Web Applications (e.g. Allmon3)","text":"

    Many will install one (or more) web applications that monitor/control their ASL nodes. These applications can provide access to many users on the web and those users may have no relation to those who setup/administer/use your systems. As such, these applications typically maintain their own list of users and associated passwords.

    Please refer to the web application manual for information on how to manage who is allowed access to the web pages.

    Also, as mentioned above, many of these applications use AMI to interact with your Asterisk and your nodes. The web application manual should have information on where to specify the AMI credentials.

    "},{"location":"user-guide/pi-detailed/","title":"Raspberry Pi Install","text":"

    The Raspberry Pi image includes the following to make a complete AllStarLink system:

    You must use the Raspberry Pi Imager even if you prefer another imager and carefully follow the instructions below. Failure to do so will result in a system that won't boot or that you will not be able to login to.

    You will install an image on a microSD card and go. This is the simplest install. For most nodes, the menus will walk you through the setup. Note: a minimum SD card size of 4G is required and it is recommended to use at least an 8G SD card.

    "},{"location":"user-guide/pi-detailed/#pi-appliance-requirements","title":"Pi Appliance Requirements","text":"

    The following are the system requirements for the Pi Appliance

    Required Recommended Hardware Arm v8 64-bit CPU Raspberry Pi 3, 4, 5, Zero 2 W, 400 Raspberry Pi 4B or 5B Memory Minimum 512M 4G Storage Minimum 4G or larger 8GB Class 10

    The ASL3 Appliance been tested on Raspberry Pi 3, 4, 5, and Zero 2W.

    "},{"location":"user-guide/pi-detailed/#step-by-step-pi-appliance-setup","title":"Step-by-Step Pi Appliance Setup","text":"

    These directions are specific for Windows by example , but in general, should work the same for the same tool set on MacOS and Linux. For each screenshot below, click on the image for a larger version.

    Note: For this example and all documentation, the hostname used shall be node63001 and the node number is 63001. Replace \"63001\" with your node number as you proceed through these directions.

    1. If you do not already have it installed, install the Raspberry Pi Imager.

    2. Download the latest release image. This will be named allstar3-arm64-X.Y.Z.img.xz where \"X.Y.Z\" is the version. The version will be something like 3.0.10. Save it to your local Downloads directory.

    3. Launch Raspberry Pi Imager from the start menu.

    4. Click on CHOOSE DEVICE and choose the type of Pi hardware on which AllStarLink 3 Appliance is being installed.

    5. Click on CHOOSE OS. Scroll to the bottom of the list that appears and select Use custom.

    6. A Select image dialog box will appear. Navigate to the Downloads folder and select the image downloaded in step 2. It will be named something like allstar3-arm64-3.0.1.img.xz. Then click Open.

    7. Connect the SD card or the SD card in a USB adapter to the computer. Then click on CHOOSE STORAGE. An option will be presented named something such as \"Mass Storage Device USB Device - 16.0 GB\". Click on that entry.

    8. Click NEXT

    9. A box labelled \"Use OS customisation?\" will appear. Click on EDIT SETTINGS

    10. Check the box next to Set hostname and enter a hostname for the device to appear as on the network. In this example, node63001 is used. Unless you have another naming convention for AllStarLink nodes, use the nodeNNNNN format.

    11. Check the box next to Set username and password and enter a username and password. It is recommended to set the username to 'asl' if you have no strong feeling about the username. Choose a good password and record the password somewhere safe. NOTE: There is NO DEFAULT USER for the image. Failure to set an account will require a reimage of the SD card with the proper settings.

    12. If the device will be connected to WiFi, check the box \"Configure wireless LAN\" and then enter the name of the wireless network in SSID and the password for that network in Password. Change the \"Wireless LAN country\" to US or another country as appropriate. Setting this is not necessary if the Pi will use a network cable rather than wireless.

    13. Check the box next to Set locale settings and change the \"Time zone\" to the desired region. The timezone settings will control any time-based announcements and schedule jobs. After completing steps 10-13, the customization should look something like:

    14. Click SAVE in \"OS Customisation\".

    15. Click YES for \"Use OS customisation?\"

    16. Click YES to continue to write the image to the SD card.

    17. Depending on the speed of the computer and the type of SD card one will have time for a beverage of their choice. When the write is complete, eject the card, remove the card from computer and insert it into the Pi. If using a USB adapter for the SD card, remove the SD card from the adapter and insert the card into the Pi. The Pi will not use the SD card in the USB adapter.

    18. Power on the Pi. Wait approximately 2 to 5 minutes for the Pi to boot and perform the various first boot tasks. Depending on the type of a Pi board used, this process could be lengthy. If there is a screen connected to the Pi, it will be noted that Pi will reboot several times. This is normal and expected.

    19. (Optional) Network connectivity may be tested using the command to ping the hostname set in step 10. For example, if the hostname set was \"node63001\" then do ping -4 node63001.local from a command prompt or PowerShell window. When you get a response, the host is connected to the network.

    20. Open the browser on your computer and navigate to https://nodeNNNNN.local. In this example the name was node63001 then one would enter https://node63001.local. Your Pi and your computer must be on the same LAN for this to work. Use the IP address rather than the hostname if the Pi is on a remote network or the hostname is unreachable. Depending on your network and browser, one may have to enter the name several times until the page displayed. The first connection will report a message that \"Your connection isn't private\". For the Pi appliance, this is acceptable. Click on Advanced and then Continue to node63001.local (unsafe). This only must be done the first time.

    21. The AllStarLink Launchpad will appear. Clicking on Web Admin Portal will take one to the web-based administration portal that includes a web-based SSH client for using asl-menu and other commands. Clicking on Node Links will take one to the Allmon3 interface. Clicking on ASL Manual will take one to the top of this manual.

    22. To setup the new node, click on Web Admin Portal. Similar to step 20, click Advanced and Continue to node63001.local (unsafe) to see the logon box.This only must be done the first time. Enter the username and password configured during the imaging process above. Click Log in

    23. The Overview tab is displayed. Using the Web Admin Portal (Cockpit) with AllStarLink is covered in the manual. At the top, click on *Turn on administrative access

      Enter the password again when prompted and click Authenticate

    24. The web-based SSH console appears as Terminal at the bottom of the left scrollable menu. Click it and SSH console will appear:

    The Terminal window is a complete interface to AllStarLink, Asterisk, and more. No SSH client is required although use of SSH directly is fully supported.

    "},{"location":"user-guide/pi-detailed/#quick-config-via-asl-menu","title":"Quick Config via ASL-Menu","text":"

    The node may now be configured. New users or those wanting a quick start experience should use the asl-menu utility with sudo asl-menu.

    The menu will appear:

    "},{"location":"user-guide/pi-detailed/#asterisk-console","title":"Asterisk Console","text":"

    To start the asterisk console from the Terminal window enter sudo asterisk -rv.

    "},{"location":"user-guide/pi-detailed/#software-updates","title":"Software Updates","text":"

    One should also apply any latest updates and reboot the device as described in Cockpit Updates.

    "},{"location":"user-guide/pi-detailed/#allmon3-default-login","title":"Allmon3 Default Login","text":"

    The default-configured username and password combination is allmon3 / password.

    You must delete this and add a new one.

    Details on updating the login(s) and other Allmon3 settings can be found in the Allmon3 section of the manual..

    "},{"location":"user-guide/pi-detailed/#logging-out","title":"Logging Out","text":"

    When finished with the admin system, click on Session in the upper right and then choose Logout.

    "},{"location":"user-guide/pi-detailed/#node-configuration","title":"Node Configuration","text":"

    Next step is to configure the node settings. YouTuber Freddie Mac has a nice ASL3 installation and configuration video. See https://youtu.be/aeuj-yI8qrU. Also see ASL3 Menu for details.

    "},{"location":"user-guide/support/","title":"ASL3 Support","text":""},{"location":"user-guide/support/#support","title":"Support","text":"

    Primary support for all versions of ASL (including ASL3) continues to be the AllStarLink Community. Please do not use the ASL Helpdesk for technical help. Support for ASL3 will NOT be handled by the ASL Helpdesk. All Helpdesk tickets will be closed with a referral to the AllStarLink Community or GitHub.

    "},{"location":"user-guide/support/#bug-reporting","title":"Bug Reporting","text":"

    Bug reports for ASL3 and surrounding issues are best handled through GitHub \"Issues\" to the source code projects at AllStarLink GitHub. There are numerous ASL3 repositories. If you're not sure which repository to use for reporting a given issue then start at ASL3. You can also ask for guidance in the AllStarLink Community.

    When reporting an issue we ask that you be comfortable enough with Linux to edit files, restart services, and perhaps copy a potentially fixed file onto your system.

    Note: to raise issues about ASL3 problems and interact with the developers you must create (or use an existing) GitHub account.

    "},{"location":"user-guide/support/#enhancement-requests","title":"Enhancement Requests","text":"

    Obviously, we cannot promise anything but we do want to hear your ideas. Please file a GitHub \"Issue\" :-)

    "},{"location":"user-guide/support/#contributing","title":"Contributing","text":"

    Have you identified a problem and have a solution? We welcome your help. Please file a GitHub issue with all the details. Even better, create a GitHub \"Pull request\" for us to review.

    "},{"location":"user-guide/troubleshooting/","title":"Basic Troubleshooting Tips","text":"

    When asking for help on AllStarLink Community or in a GitHub Issue here are some tips on how to ask for help and how to get common troubleshooting information that is often needed to understand your specific issues.

    "},{"location":"user-guide/troubleshooting/#tips-on-asking-for-help","title":"Tips on Asking for Help","text":"

    Keep in mind that when asking for help, the pool of people likely able to help have no understanding of your personal setup, situation, skills, etc. In order to get good help, you need to ask a good question. Make sure when asking for help that you hit the following points:

    "},{"location":"user-guide/troubleshooting/#explain-precisely-what-is-not-working-properly","title":"Explain Precisely What is not Working Properly","text":"

    Requests for help such as \"ASL doesn't work\" or \"my pi crashes\" or \"I can't do X\" without any further details or information is unable to be diagnosed and assisted.

    Examples of GOOD ways to ask a question:

    Basically, before submitting your question to Community, make sure your post contains the basics of \"What am I trying to do\", \"what do I think should be happening\", and \"What is happening instead\". If you are filing an issue on GitHub, use the Issue Templates - they ask for information for a reason.

    "},{"location":"user-guide/troubleshooting/#be-ready-to-provide-requested-basic-troubleshooting-info","title":"Be Ready to Provide Requested Basic Troubleshooting Info","text":"

    If you are going to ask for help, be ready and willing to provide the basic troubleshooting information that is requested. For example, if you're having a problem linking to another node, be ready to provide - or even better provide in advance - the basic troubleshooting information needed to help with your problem.

    Items that may be requested include elements such as:

    "},{"location":"user-guide/troubleshooting/#allstarlink-does-not-support-certain-other-projectstools","title":"AllStarLink Does Not Support Certain Other Projects/Tools","text":"

    Core AllStarLink does not support certain other projects/tools such as HamVOIP, Supermon/Supermon2, SkyWarnPlus, or AllScan. Some of the developers of these tools such as SkyWarnPlus and AllScan are active and answer questions on the Community and others are not. If you are told, \"that isn't supported but have you tried doing it in this supported way\", consider giving it a try.

    "},{"location":"user-guide/troubleshooting/#allstarlink-v3-contains-14-years-of-changes","title":"AllStarLink v3 Contains 14+ Years of Changes","text":"

    The release of AllStarLink v3 uses modern Asterisk 20 LTS. Legacy installs use Asterisk 1.4 with was end of life in 2012. Thus, there are literally 14+ years of changes of how Asterisk operates that app_rpt (the main engine behind ASL) had to conform to. Notably, the configuration syntax which changed gradually for time for users staying current with Asterisk is an \"all of a sudden\" change in ASL3. One cannot paste old configuration into modern ASL3 and have it \"just work\".

    Additionally, Asterisk no longer runs as the root user on Linux which means that its ability to do things to the system such as make OS changes is purposefully restricted. ASL3 and Asterisk are very security conscious as many of our users run ASL3 systems on the open Internet.

    "},{"location":"user-guide/troubleshooting/#give-questioners-the-benefit-of-the-doubt","title":"Give Questioners the Benefit of the Doubt","text":"

    In general, if people are asking questions about the details of your problem, they are likely trying to help and not nitpick. In general, if many people are having the problem. it is already known and likely documented. If it's not, then likely your issue is unique to your setup and providing the requested troubleshooting is needed to help.

    "},{"location":"user-guide/troubleshooting/#basic-troubleshooting","title":"Basic Troubleshooting","text":""},{"location":"user-guide/troubleshooting/#gathering-logs","title":"Gathering Logs","text":"

    If you are asked to provide logs, there are two standard ways to provide logs - using Cockput and using the journalctl tool from the command line.

    "},{"location":"user-guide/troubleshooting/#gathering-logs-with-cockpit","title":"Gathering Logs with Cockpit","text":"

    In the Cockpit interface, navigate to Services on the left navbar and click. The service list will display. Wait for the full list to load and then click on the blue link name of the service. The common services are:

    After clicking on one of the links wait until Service logs fills in and then view the logs. Clicking View all logs will show more logs.

    "},{"location":"user-guide/troubleshooting/#gathering-logs-with-clissh","title":"Gathering Logs with CLI/SSH","text":"

    Use the journalctl -xeu command to provide logging output. For example:

    journalctl -xeu asterisk.service\n

    Common services are: * allmon3.service - Allmon 3

    "},{"location":"user-guide/troubleshooting/#gathering-configs","title":"Gathering Configs","text":"

    It is often needed to gather certain configuration details. Use the Cockpit Console or the SSH/CLI interface. People asking for information will likely provide the file needed in the request. However here are some easy ways to get the data requested:

    When possible, please copy/paste text and not screenchots into systems for troubleshooting. Text is far easier to deal with.

    "},{"location":"user-guide/updates/","title":"Updating ASL3","text":"

    All of the software components that make up AllStarLink v3 are provided by software packages (Debian .deb files) from an centralized Apt repository. This makes upgrades quick and painless for users.

    "},{"location":"user-guide/updates/#standard-debian-upgrades","title":"Standard Debian Upgrades","text":"

    Whether AllStarLink v3 is installed on a standard Debian 12 installation or you are using the Pi appliance image, updates follow the standard Debian format:

    sudo apt update\nsudo apt upgrade -y\n

    This will update all AllStarLink v3 software as well as the underlying operating system. AllStarLink will never release any update that is a \"breaking change\" through the apt system that doesn't have either an automated upgrade process or a well-documented manual step-by-step process.

    A reboot is only needed when a new Linux kernel has been installed.

    Sometimes an update may ask what to do with a configuration file. It is generally best to answer such questions with N meaning \"keep your currently-installed version. Such a question will look like:

    Configuration file '/etc/asterisk/rpt.conf'\n ==> Modified (by you or by a script) since installation.\n ==> Package distributor has shipped an updated version.\n   What would you like to do about it ?  Your options are:\n    Y or I  : install the package maintainer's version\n    N or O  : keep your currently-installed version\n      D     : show the differences between the versions\n      Z     : start a shell to examine the situation\n The default action is to keep your current version.\n*** rpt.conf (Y/I/N/O/D/Z) [default=N] ? N\n

    AllStarLink will clearly announce when there is a change that may break existing configurations.

    If you want updates to install without any questions being asked specify the DEBIAN_FRONTEND=noninteractive variable:

    DEBIAN_FRONTEND=noninteractive sudo apt update\nDEBIAN_FRONTEND=noninteractive sudo apt upgrade -y\n

    Using this command will automatically choose \"N\" to any question about configuration - i.e. it will always keep your installed version.

    "},{"location":"user-guide/updates/#note-on-dahdi","title":"Note on DAHDI","text":"

    In the past, managing the DAHDI kernel module needed by app_rpt had been problematic during kernel upgrades. Previously, the recommendation had been to never upgrade the kernel. This advice is no longer valid and should not be followed for security and practicality reasons.

    AllStarLink v3 automatically manages and rebuilds the DAHDI kernel modules during software updates using the DKMS system. After an upgrade, rebooting into a new version of the Linux kernel will \"Just Work\".

    "},{"location":"user-guide/whats-new/","title":"What's New?","text":"

    Ready to get on with installation? We thought so, but first consider that ASL3 is very new. You'll probably want to install on removable media or on a non-production system. You probably don't want to install a production system right away. Give yourself some time to test that everything works as you'd expect.

    Note: if you have installed an earlier version of ASL you need to know that ASL3 has many changes and that there's a bit of a learning curve. For those who have used ASL2 or other versions of AllStar you should be aware of a few important details:

    The two node numbers per user policy remains in effect. Node numbers are a limited resource. Please make a best effort to use new media, a new server, or NNX to build a node for testing. We do make exceptions on a case by case basis but we'd really appreciate it if you could use a resource you already have.

    "},{"location":"user-guide/whats-new/#menu-changes","title":"Menu Changes","text":"

    The ASL3 menu has been updated. Our goal is to provide an easy way for you to configure your node and to minimize the need to edit any configuration files.

    At the same time, we have removed most of the \"OS\" configuration options found in earlier implementations. Historically, ASL has been installed on many OS variants and the OS commands that work on one version do not always work on others. For that reason there are no OS related commands in the ASL3 menu.

    See Menu for details.

    "},{"location":"user-guide/whats-new/#configuration-changes","title":"Configuration Changes","text":"

    The first thing to know about ASL3 configuration is that the menu can be used to make and view common changes. Here are the highlights. You may want to read about Advanced Topics and New Commands details later.

    "},{"location":"user-guide/whats-new/#ap-measurement","title":"A&P Measurement","text":"

    Adoption and Performance measurements are to help ASL understand growth and technical aspects of ASL3. No sensitive or private information is collected. For details see GitHub/AllStarLink/ASL3-Usage. Reports are available at repo.allstarlink.org/reports/.

    "}]} \ No newline at end of file +{"config":{"lang":["en"],"separator":"[\\s\\-]+","pipeline":["stopWordFilter"]},"docs":[{"location":"","title":"Welcome to AllStarLink 3","text":"

    AllStarLink\u2019s version 3 is the next generation of AllStar repeater and hotspot software. This version of AllStar has been redesigned to run on Asterisk 20 LTS (Long Term Support), the latest Debian Linux operating system, and modern hardware or virtual machines. The update from Asterisk 1.4 to Asterisk 20 implements over 15 years of Asterisk bug fixes, security improvements and enhancements. It brings with it the latest Asterisk applications, channel drivers and additional functionality.

    This update required app_rpt (the Asterisk application that is AllStar) and it's various modules to be heavily modified. Many memory leaks have been addressed, modules load or refresh more reliably and many bugs have been squashed. This all adds up to improved stability and uptime.

    In addition to the many app_rpt improvements and fixes, the code base is easier to maintain and enhance. The goal being to make app_rpt code accessible to more AllStar developers. Also, the app_rpt code base has been modified to meet Asterisk\u00ae coding guidelines.

    Supported Hardware & Software

    "},{"location":"#new-features","title":"New Features","text":"

    In addition to the app_rpt code update, we have added many new features to make ASL3 the best AllStar release yet.

    "},{"location":"adv-topics/allowdenylists/","title":"Allow and Deny Lists","text":"

    These changes can easily be made from the ASL3 menu. Here's how to change and view the lists at the Asterisk CLI should you choose to do so. These lists were previously referred to as whitelist and blacklist.

    ASL3 provides a way to establish either an \"allowlist\" or a \"denylist\" to limit access to your node(s). New to ASL3 is the capability for nodes on the same server to have different lists. Unlike ASL2, no modifications or scripts are required to the configuration files and there is no need to restart or reload the server. All changes are made to the Asterisk database. As soon as you make a change it takes effect.

    The \"allowlist\" allows inbound connects and blocks all others. Nodes on the same server are always allowed. If any nodes are on the \"allowlist\" the \"denylist\" is ignored.

    The \"denylist\" prevents nodes from connecting to your node. Add a \"denylist\" if you only have a few nodes you want to block.

    To see what's in the database.

    To delete an entire \"allowlist\"

    "},{"location":"adv-topics/audio-level/","title":"Calibrating Audio Levels For The Allstar Link System","text":""},{"location":"adv-topics/audio-level/#introduction","title":"Introduction","text":"

    Proper audio level calibration is crucial for ensuring good audio quality, and consistent audio levels from node-to-node. Fortunately, the procedure is not that complicated. To properly perform the audio level calibration, you will need the following items:

    "},{"location":"adv-topics/audio-level/#assumptions","title":"Assumptions","text":""},{"location":"adv-topics/audio-level/#procedure","title":"Procedure","text":""},{"location":"adv-topics/audio-level/#checking-the-radios-transmitters-peak-deviation","title":"Checking the Radio's Transmitter's Peak Deviation","text":"

    If you intend to drive the transmitter at the microphone input, or at another audio input before pre-emphasis and limiting, you must ensure that the peak deviation of the transmitter is no more than what is required to operate on the channel (typically 5KHz). If you driving the modulator directly from a URI, this section can be skipped.

    1. Disable the generation of CTCSS tones in the transmitter. (If you don't encode CTCSS, skip this step and steps 6 and 7)
    2. Hook up the audio signal generator to the transmitter microphone input, set it for a 1kHz sine wave, and the lowest output setting. Hook up the service monitor to the transmitter RF output (with appropriate attenuation for protection of your Service Monitor) and key the transmitter.
    3. On the service monitor, monitor the peak deviation of the transmitter. Gradually increase the output of the audio signal generator until further adjustment does not increase the deviation. Make a note of the peak deviation at this point.
    4. If the peak deviation is more than 5kHz, then adjust the transmitter deviation pot or soft pot so that the peak deviation reading on the service monitor is no more than 5kHz. Check the shape of the audio waveform on the service monitor's scope. It should show evidence of hard limiting (it should look like a square wave with rounded edges)
    5. Reduce the level of the audio signal generator so that the service monitor reads 3kHz of peak deviation. The audio waveform should now be a sine wave with no evidence of limiting.
    6. Re-enable the generation of CTCSS tones in the transmitter
    7. Check that the peak deviation is no more than 3.6 kHz. If it is more than this, adjust the CTCSS level so that a reading of 3.6kHz is attained.
    "},{"location":"adv-topics/audio-level/#adjusting-the-transmit-audio-level","title":"Adjusting the Transmit Audio Level","text":"
    1. Disable CTCSS tone generation in the transmitter or at the CTCSS encoder.
    2. Enable the test tone generator in rpt.conf and restart Asterisk
    3. Connect the radio interface to the transmitter
    4. Bring up the Asterisk CLI.
    5. From the asterisk CLI, type rpt fun yournodenumber *989 where yournodenumber is the node number assigned to you.
    6. The transmitter should key and you may hear a tone. Adjust the TX level until the service monitor indicates 3kHz of peak deviation.
    7. From the asterisk CLI, type rpt fun yournodenumber #. This will kill the test tone and unkey the transmitter.
    8. If encoding CTCSS, re-enable the CTCSS encoder in the transmitter, then repeat the previous 3 steps. You should see no more than 3.6KHz of deviation with the CTCSS tone generator enabled. If it is higher than this, adjust the CTCSS level per your transmitter's service documentation until a 3.6KHz of deviation is seen on the service monitor.
    "},{"location":"adv-topics/audio-level/#adjusting-the-receive-audio-level","title":"Adjusting the Receive Audio Level","text":"

    The procedure depends on what type of radio you have interfaced. If your radio is a repeater, use the full duplex procedure, if your radio is a remote base or simplex node, use the half-duplex procedure. Note: if you have a URI, the automated setup procedure should have set the receive level very close to what it needs to be. This section can be used to verify the receive level is correct for a URI, but the URI receive level should always be set using the automated adjustment procedure.

    "},{"location":"adv-topics/audio-level/#full-duplex-radios","title":"Full Duplex Radios","text":"

    Adjusting the receive level of a full duplex radio is easy.

    1. Disable the transmitter CTCSS encoder. Bypass the receiver CTCSS decoder so that it is in carrier squelch.
    2. Monitor the deviation of the transmitter on the service monitor while simultaneously injecting a 1kHz tone frequency at 3kHz deviation on the receiver.
    3. Adjust the receive level pot until you see 3kHz of deviation on the transmit frequency.
    4. Disable the signal on the receiver. The receive audio level is now set.
    "},{"location":"adv-topics/audio-level/#half-duplex-radios","title":"Half Duplex Radios","text":"

    Half duplex radios will require the use of a second node which has had the transmit level properly set, and is not transmitting a CTCSS tone. The second node must be receivable by your service monitor or deviation meter but does not need to be co-located with the node undergoing the receive level adjustment. The CODEC used to send audio to the second node must either be uLAW or ADPCM. The GSM codec will give inaccurate results.

    1. Using the Asterisk CLI, connect the node to be adjusted to the transmit node: rpt fun yournodenum *3txnodenum
    2. Monitor the deviation of the second node's transmitter on the service monitor while simultaneously injecting a 1kHz tone frequency at 3kHz deviation on the receiver on the first node.
    3. Adjust the receive level pot until you see 3kHz of deviation on the transmit frequency.
    4. Disable the signal on the receiver. The receive audio level is now set.
    5. Disconnect the receive node from the transmitter: rpt fun yournodenum *1txnodenum
    "},{"location":"adv-topics/autopatch/","title":"Autopatch","text":"

    The autopatch feature in app_rpt allows users on the radio to interconnect with the public switched telephone network.

    "},{"location":"adv-topics/autopatch/#regulatory-issues","title":"Regulatory Issues","text":"

    Some countries do not allow third-party traffic through autopatching or telephone interconnection on amateur radio frequencies. In addition, if a system is linked between one country which allows autopatching and one which doesn't, just the passing of the traffic itself across the link could be considered a violation of the rules in the prohibiting country. In countries which permit autopatching, users need be made aware of this and should take the node off-link before using the autopatch.

    "},{"location":"adv-topics/autopatch/#security-issues","title":"Security Issues","text":"

    The examples provided here do not secure the autopatch against toll fraud. A lot can be done to prevent or reduce toll fraud, but the prevention measures vary by where you are located geographically. Most of the prevention measures will be in extensions.conf. You can write contexts to reject certain number sequences. The asterisk book should be consulted to see how to do this, Before placing any autopatch in service, please make sure you have secured your system so that toll fraud does not become an issue.

    "},{"location":"adv-topics/autopatch/#selecting-a-voip-provider","title":"Selecting a VOIP Provider","text":"

    Call termination into the public switched telephone network is a service offered by a Voice Over IP provider (VOIP). This is a highly competitive business, and there are lots of VOIP providers offering termination for Asterisk users. Termination is usually provided using the SIP protocol, while IAX is offered by a few providers.

    When selecting an VOIP provider, make sure they provide setup instructions with example configurations including usernames, and passwords. Quite a few VOIPs will automatically generate a custom SIP or IAX stanza for insertion into sip.conf, or iax.conf respectively.

    "},{"location":"adv-topics/autopatch/#configuration-files","title":"Configuration Files","text":"

    Four configuration files will be used to setup the autopatch: rpt.conf, extensions.conf, modules.conf, and iax.conf or sip.conf. The VOIP provider should provide a stanza for you to insert in iax.conf or sip.conf. The configuration below shows how each of these files are used when an outgoing autopatch call is made.

    In the modules.conf file, add these modules at the bottom of the file, but above the [global] stanza:

    load => bridge_builtin_features.so\nload => bridge_builtin_interval_features.so\nload => bridge_holding.so\nload => bridge_native_rtp.so\nload => bridge_simple.so\nload => bridge_softmix.so\nload => chan_bridge_media.so\n

    In the rpt.conf configuration file, a context in extensions.conf is defined in a node stanza using the context key:

    [1234]\n\ncontext = autopatch                 ; dialing context for phone\ncallerid = \"Repeater\" <0000000000>  ; callerid for phone calls\naccountcode = RADIO                 ; account code (optional)\n
    Also in the rpt.conf file, entries are added to the function stanza so that users can bring up or take down the autopatch: 
    [functions]\n61 = autopatchup,noct=1,farenddisconnect=1,dialtime=20000,context=autopatch\n62 = autopatchdn \n

    In extensions.conf, a stanza for the autopatch context is added which refers to the peer stanza in iax.conf or sip.conf.

    For outbound IAX connections extensions.conf 

    [autopatch] \n\nexten => _1NXXNXXXXXX,1,Dial,IAX2/peername/${EXTEN} \nexten => _1NXXNXXXXXX,2,Congestion \n
    For outbound SIP connections it should look like this: 
    [autopatch] \n\nexten => _1NXXNXXXXXX,1,Dial,SIP/peername/${EXTEN} \nexten => _1NXXNXXXXXX,2,Congestion \n

    The peer stanza you insert in iax.conf or sip.conf should preferably be provided by your VOIP provider. You'll want to use the stanza they identify for performing outgoing connections. The SIP module has been removed from ASL3 and one must now use PJSIP. A nonoperational example of an iax2 peer stanza looks like this: 

    [peername]\ntype=peer \nhost=127.0.0.1 \nsecret=nunya\nauth=md5 \ndisallow=all \nallow=gsm \nallow=ulaw\n

    "},{"location":"adv-topics/autopatch/#autopatch-options","title":"Autopatch Options","text":"

    There are several options that can be passed to the autopatchup command class.

    This table summarizes what they do:

    Autopatch Option Description context Override the context specified for the autopatch in rpt.conf dialtime The maximum time to wait between DTMF digits when a telephone number is being dialed. The patch will automatically disconnect if this time is exceeded. The value is specified in milliseconds. farenddisconnect When set to 1, the patch will automatically disconnect when the called party hangs up. The default is send a circuit busy tone until the radio user brings the patch down. noct When this is set to 1, the courtesy tone during an autopatch call will be disabled. The default is to send the courtesy tone whenever the radio user unkeys. quiet When set to 1, do not send dial tone, voice responses, just try to connect the call. voxalways When set to 1, enables vox mode. exten Overrides the default autopatch extension. nostar Disables the repeater function prefix"},{"location":"adv-topics/autopatch/#simplex-autopatch","title":"Simplex Autopatch","text":"

    For nodes with duplex set to 0 or 1, Simplex Autopatch operation is supported. This is accomplished by using a VOX on the audio coming from the telephone call, because of the half-duplex nature of the node. To avoid having audio from the telephone call keep the transmitter engaged for extended periods of time (if there is some source of continuous audio), there are two time-out values.

    Vox timeout, which is the maximum amount of time that the VOX can hold the transmitter engaged, which by default is 10 seconds. The other one is the Vox recovery time, which is the amount of time that the Vox is disabled during times of continuous audio, after the Vox timeout, which is 2 seconds. During continuous audio, it transmits for 10 seconds, then stops for 2 seconds (so that you can transmit to it and perhaps disconnect the call), then transmits for another 10 seconds, then stops for 2 seconds. This repeats until the call is disconnected.

    Additionally, to prevent 'chop-off' of the first syllable or two of the audio from the telephone call, the audio is delayed to allow for the transmitter to start transmitting and any CTCSS tones to be decoded. This delay is typically 500ms, but can be adjusted using the 'simplexpatchdelay' parameter. It is specified in units of 20 milliseconds.

    These time-out values may be overridden by using the following configuration parameters located in the [node-number] section of rpt.conf: 

    [1999]\nvoxtimeout = 10000 ; vox timeout time in ms\nvoxrecover = 2000  ; vox recover time in ms\nsimplexpatchdelay = 25 ; Delay for transmit while in patch in 20ms increments\nsimplexphonedelay = 25 ; Delay for phone while in patch for 20ms increments\n

    "},{"location":"adv-topics/broadcastify/","title":"Streaming Node Audio to Broadcastify","text":"

    To broadcast your node's audio on Broadcastify, you will need a Broadcastify account. You can then apply for a feed. This link provides information on applying for a feed.

    After you have your account and feed credentials, you a ready to setup AllStar.

    "},{"location":"adv-topics/broadcastify/#setup-a-feed","title":"Setup a Feed","text":"

    The above parameters have these meanings: 

    -- preset cbr 16` = use constant bit rate 16\n-r = Assume the input file is raw pcm\n-m m = Mode mono\n-s 8 = sample rate 8\n--bitwidth 16 = bit width is 16 (default)   \n

    If you experience any problems, look at /var/log/ezstream.log for error messages.

    "},{"location":"adv-topics/broadcastify/#migrating-an-existing-feed","title":"Migrating an existing feed","text":"

    If you have an existing feed, you will need to upgrade your existing xml file to the new format. You can use the following commands. 

    cd /etc\nezstream-cfgmigrate -0 ezstream.xml > ezstream.xml.new\ncp ezstream.xml ~/\nmv ezstream.new ezstream.xml\n

    "},{"location":"adv-topics/cloud-kernel/","title":"Using AllStarLink/app_rpt with Cloud Kernels","text":"

    Many virtual hosts or VPS providers leverage the Debian \"cloud kernel\" version of the Debian Linux OS. These providers include Amazon AWS, Microsoft Azure, Linode, Vultr, DreamHost, and DigitalOcean. Using their stock Debian 12 installation image comes with the \"cloud kernel\" installed. While this kernel is usually beneficial due to its smaller footprint than the full kernel through the removal of hardware elements, the Asterisk app_rpt module requires the support of these elements.

    "},{"location":"adv-topics/cloud-kernel/#diagnosing-a-dahdi-kernel-issue","title":"Diagnosing a DAHDI Kernel Issue","text":"

    Problems with Asterisk/app_rpt related to the DAHDI module being unavailable can be see by the following log messages at boot:

    [2024-12-10 15:09:19.780] ERROR[930155] loader.c: res_timing_dahdi declined to load.\n[2024-12-10 15:09:19.818] NOTICE[930201] app_rpt.c: Normal Repeater Init  63001\n[2024-12-10 15:09:19.818] WARNING[930220][C-00000001] chan_dahdi.c: Unable to open '/dev/dahdi/pseudo': No such file or directory\n[2024-12-10 15:09:19.818] ERROR[930220][C-00000001] chan_dahdi.c: Unable to dup channel: No such file or directory\n[2024-12-10 15:09:19.818] ERROR[930220] app_rpt/rpt_bridging.c: Failed to request dahdi/pseudo\n[2024-12-10 15:09:19.818] WARNING[930220] app_rpt.c: Disabled broken repeater 63001\n

    This appears in /var/log/asterisk/messages.log not long after Asterisk begins to start.

    "},{"location":"adv-topics/cloud-kernel/#converting-from-cloud-to-normal-kernel","title":"Converting from Cloud to Normal Kernel","text":"

    After installing a Debian Linux 12 image that is using a cloud kernel, the following procedure can be used to change the OS to use the normal kernel that will allow the building of the DAHDI module needed for app_rpt.

    Now setup of ASL3/app_rpt should be able to continue.

    "},{"location":"adv-topics/commands/","title":"New Commands","text":"

    New and different ASL3 commands are mentioned here. Older ASL commands are still documented on the AllStarLink Wiki

    "},{"location":"adv-topics/commands/#app_rpt-commands","title":"app_rpt commands","text":" 
    [general]\nnode_lookup_method = both   ;method used to lookup nodes\n                            ;both = dns lookup first, followed by external file (default)\n                            ;dns = dns lookup only\n                            ;file = external file lookup only\n
    "},{"location":"adv-topics/commands/#chan_simpleusb-commands","title":"chan_simpleusb commands","text":"

    susb show settings is used to show the currently selected node's settings. This replaces susb tune.

    "},{"location":"adv-topics/commands/#chan_usbradio-commands","title":"chan_usbradio commands","text":"

    radio show settings is used to show the currently selected node's settings. This replaces radio tune.

    "},{"location":"adv-topics/commands/#simpleusb-and-usbradio-tune-menus","title":"SimpleUSB and USBRadio Tune Menus","text":"

    The simpleusb-tune-menu and radio-tune-menu utility programs have been updated with new options. The new options allow you change the operation of the respective channel driver, in realtime, without having to manually edit the simpleusb.conf or usbradio.conf files.

    In addition to the updating settings, you can now view the live status of the COS, CTCSS inputs and PTT output. This allows you to easily view and change their settings.

    "},{"location":"adv-topics/commands/#echolink","title":"Echolink","text":""},{"location":"adv-topics/commands/#echolink-commands","title":"Echolink commands","text":"

    echolink show nodes is used to view the currently connected echolink users.

    echolink show stats is used to view the channel statistics for echolink. It shows the number of in-bound and out-bound connections. It also shows the cumulative system statistics, along with the statistics for each connected nodes.

    "},{"location":"adv-topics/commands/#helper-scripts","title":"Helper scripts","text":""},{"location":"adv-topics/commands/#debugging","title":"Debugging","text":"

    Previously app_rpt and associated channels supported setting the debug level with an associated app / channel command. These app / channel commands have been removed and replaced with the asterisk command:

    core set debug x module Where x is the debug level and module is the name of the app or module.

    Examples:

    "},{"location":"adv-topics/commands/#usb-eeprom-operation","title":"USB EEPROM Operation","text":"

    chan_simpleusb and chan_usbradio allow users to store configuration information in the EEPROM attached to their interface(s). The CM119A can have manufacturer information in the same area that stores the user configuration. The CM119B does have manufacturer data in the area that stores user configuration. The manufacturer data cannot be overwritten. The user configuration data has been moved higher in memory to prevent overwriting the manufacturer data. If you use the EEPROM to store configuration data, you will need to save it to the EEPROM after upgrading. Use susb tune save or radio tune save.

    "},{"location":"adv-topics/conftmpl/","title":"Asterisk Templates","text":"

    Asterisk now has a template configuration option. This can make editing Asterisk conf flies less tedious. So far, ASL3 is using templated configurations in the \"rpt.conf\", \"simpleusb.conf\", and \"usbradio.conf\" files.

    "},{"location":"adv-topics/conftmpl/#etcasteriskrptconf","title":"/etc/asterisk/rpt.conf","text":"

    The \"rpt.conf\" file remains basically the same as it was in ASL2. But, the organization has changed to make it easier to maintain, particularly for multi-node systems.

    The template for a node in rpt.conf is named \"[node-main]\". Every node tagged with \"(node-main)\" inherits all the template settings. Template settings are overwritten by nodes with \"(main-node)\" attached.

    As you will see, adding a new node requires the addition of just a few lines. For example adding three nodes to your system may need nothing more than:

    [1999](node-main)\nrxchannel = SimpleUSB/1999\nidrecording = |iWB6NIL\nstartup_macro = *32000\n\n[1998](node-main)\n;this node would use the ID in the template\nrxchannel = Radio/1998\nmorse = morse_1998\n\n[1997](node-main)\n;This might be a hub node if \"rxchannel = dahdi/pseudo\" is in the template\n
    "},{"location":"adv-topics/conftmpl/#etcasterisksimpleusbconf-etcasteriskusbradioconf","title":"/etc/asterisk/simpleusb.conf, /etc/asterisk/usbradio.conf","text":"

    The \"/etc/asterisk/simpleusb.conf\" and \"/etc/asterisk/usbradio.conf\" files have also adopted a templated configuration. Again, the template for a node is named \"[node-main]\". Every node tagged with \"(node-main)\" inherits all the template settings. Template settings are overwritten by nodes with \"(main-node)\" attached.

    "},{"location":"adv-topics/conftmpl/#asl-menu-support","title":"ASL menu support","text":"

    The new ASL3 menu is fully aware of the templated configuration and handles adding, updating, and removing nodes.

    "},{"location":"adv-topics/hotspot_hd/","title":"Hot Spot Unkey Delay","text":"

    When setting up a hot spot select the Duplex 1 option with the asl-menu will allow the hot spot to function. However the unkey delay of the radio transmiter will be too long.

    These steps will shorting the unkey delay and will greatly improve hot spot responsiveness.

    This setup will likely be a single menu selection with the next release of ASL3.

    "},{"location":"adv-topics/httpreg/","title":"HTTP Registration","text":"

    AllStarLink is moving from IAX2 to HTTP registration. ASL3 has a new module for http registration. ASL3 menu users need not be concerned about this change. The asl3-menu sets this for you.

    The conf new file is /etc/asterisk/rpt_http_registrations.conf. The setting values are the same as they were in IAX2 registration.

    [General]\n\n[registrations]\n;Change the register => line to match your assigned node number and password.\nregister => 1999:password@register.allstarlink.org    ; This must be changed to your node number, password\n

    While IAX registration still works, please do not register with both HTTP and IAX. That would result in unnecessary server load for no gain. The long term plan is to do away with IAX registration. HTTP registration also allows load balancing and other advantages not available with IAX. The ASL3 IAX module has been replaced with the Asterisk LTS IAX2 module for upstream compatibility.

    "},{"location":"adv-topics/iaxreg/","title":"IAX-Based Registration","text":"

    This documents how to use IAX-based registration instead of HTTP-based registration with ASL3.

    Do Not Do This \"Just Because\"

    Only convert your node to using IAX-based registration because you have a known problem using HTTP registration. Use of IAX registration is significantly higher server use than HTTP registration and 99+% of all users DO NOT have a reason to use it.

    "},{"location":"adv-topics/iaxreg/#reason-to-use-iax-based-registration","title":"Reason to use IAX-Based Registration","text":"

    The following are the only reasons to use IAX-based registration with ASL3:

    1. Outbound HTTP/HTTPS is blocked but you have outbound UDP port access available. For example, your node sits behind a proxy or other security or content filtering device that breaks ASLv3 registration.

    2. Your node is behind a connections that handle HTTP traffic differently than IAX traffic. A known example is AT&T and T-Mobile hotspots. These hotspots intercept/direct web (HTTP/HTTPS) traffic and route it differently from IAX traffic. In these use cases, the HTTP-based registration will register a proxy/accelerator IP address rather than the IP being used by the node.

    "},{"location":"adv-topics/iaxreg/#using-iax-registration-instead-of-http-registration","title":"Using IAX Registration Instead of HTTP Registration","text":"

    To change from HTTP to IAX registration, do the following as root (i.e. use sudo -s):

    1. Edit /etc/asterisk/modules.conf and change the line for res_rpt_http_registrations.so from \"load\" to \"noload\": 
      noload => res_rpt_http_registrations.so\n

    2. Edit /etc/asterisk/iax.conf and add a register => line under the [general] stanza. This can be copied from a functional rpt_http_registrations.conf file: 

      [general]\nregister => 63001:VerySecret@register.allstarlink.org\n

    3. Restart asterisk with systemctl restart asterisk

    4. Run the asterisk console with asterisk -r and test the IAX registration with the command iax2 show registry: 

      node63001*CLI> iax2 show registry\nHost                  dnsmgr  Username  Perceived                 Refresh  State\n34.105.111.212:4569   Y       63001     192.0.2.171:4569          180      Registered\n

      If it lists \"State\" as \"Registered\" you're all set.

    "},{"location":"adv-topics/incompatibles/","title":"Incompatibilities with Legacy Versions & Configs","text":"

    AllStarLink v3 has certain incompatible changes with older versions of the AllStarLink system.

    "},{"location":"adv-topics/incompatibles/#voterrtcm-default-port","title":"Voter/RTCM Default Port","text":"

    Modern installations of Asterisk runs as the unprivileged asterisk user rather than as root. Standard Linux convention prohibits non-root users to listen on a TCP port below 1024. The default port for Voters/RTCMs has been changed to 1667 when previously it was 667.

    If the Voter/RTCM port cannot be easily changed, then the following configuration can be made to the underlying operating system:

    echo net.ipv4.ip_unprivileged_port_start=667 > /etc/sysctl.d/aslport667.conf\nsysctl -p\nsystemctl restart asterisk\n

    If running the AllStarLink Pi Appliance (or another system with a firewall), inbound to port 667/UDP must be permitted. For directions on how to do this with the Pi Appliance see Managing the Firewall.

    "},{"location":"adv-topics/incompatibles/#usb-udev","title":"USB udev","text":"

    A udev rule is needed to allow Asterisk running as non-root access to the USB system. ASL3 systems installed from debs, apt install or images will already have this rule in place.

    /etc/udev/rules.d/90-asl3.rules\nSUBSYSTEM==\"usb\", ATTRS{idVendor}==\"0d8c\", GROUP=\"plugdev\", TAG+=\"uaccess\"\n

    The UDev subsystem must be reloaded and then the USB device re-inserted into the port. UDev is reloaded with the command

    udevadm control --reload\n

    Alternatively, reboot the system.

    "},{"location":"adv-topics/multinodesnetwork/","title":"Multiple Nodes on the Same Network","text":"

    Multiple AllStarLink nodes on the same IP subnet (i.e. your \"local network\") when those nodes are behind a NAT device can be a challenge. The reason is that AllStarLink works exclusively with public IP addresses. When two nodes are on the same network behind a NAT/router they independently need to use a public IP address with the AllStarLink system but need to know about each other's local IPs rather than their shared public IP. Additionally both nodes cannot use the same UDP port as they will fight over it with the NAT/router.

    For this example, let's consider two nodes on the same subnet. The nodes are:

    Node Local IP 630010 192.168.0.10 630011 192.168.0.11

    These nodes are behind a NAT/router with a public IP address of 203.0.113.186.

    When each node connects and registers with AllStarLink, its perceived registration IP address will be 203.0.113.186. If another node asks \"where is node 630010\" it would get back the answer 203.0.113.186. However, this is also the case for the internal nodes. If node 630011 asks \"where is node 630010\" it also gets back 203.0.113.186. For various network reasons, this will not work.

    One also has to consider the UDP port. By default, each Asterisk system uses the default UDP port of 4569. This is a published standard. However, through a NAT device, only one system can use the port at a time. Thus our local nodes 630010 and 630011 will fight over that port.

    The make two (or more) nodes on the same network function, the following steps must be taken:

    1. Assign each node's server a different IAX port
    2. Tell each node's server about the other node's internal address and port

    In this example, node 630011 will be assigned a different UDP port. The table then of the nodes is:

    Node Local IP Port 630010 192.168.0.10 UDP 4569 630011 192.168.0.11 UDP 4570

    This document assumes one knows how to forward the port internally for one's router/NAT device and that is has already been done.

    "},{"location":"adv-topics/multinodesnetwork/#assigning-a-different-iax-port","title":"Assigning a Different IAX Port","text":"

    There are two places where the IAX port must be changed when using a non-standard port:

    1. /etc/asterisk/iax.conf
    2. The AllStarLink Portal for that node's server
    "},{"location":"adv-topics/multinodesnetwork/#changing-iaxconf","title":"Changing iax.conf","text":"

    The best way to edit the IAX port is to use asl-menu. Run sudo asl-menu. Choose 1 Node Settings and then 4 Update Asterisk IAX port. Set the new UDP port in the box and hit OK. Then choose 2 Restart Asterisk. Then close asl-menu.

    Alternatively, edit the file directly. As root (i.e. sudo -s), edit the file /etc/asterisk/iax.conf. Find the line:

    bindport = 4569\n

    and change it to a different port. The ASL3 appliance comes preconfigured to permit traffic on any port between 4560 and 4580. It is recommended to choose a port in this range for any alternative IAX port. In this example, the node 630011 will be changed to port 4570. Edit the file to state:

    bindport = 4570\n

    and save and close the file. Restart asterisk with systemctl restart asterisk.

    "},{"location":"adv-topics/multinodesnetwork/#changing-the-portal","title":"Changing the Portal","text":"

    Server-Node Relationship

    Following the operating model, the base Asterisk/app_rpt installation is the \"Server\" and a \"Server\" hosts one or more \"Nodes\". Make sure that each node is assigned to a different server if the nodes are configured on different installations.

    Do the following to change a node's server's IAX port.

    1. Logon to https://www.allstarlink.org using your AllStarLink username and password.

    2. Click on Portal and then Node Settings.

    3. In the table, find the Server for the node you want to edit. In this example node 630011 is located on server node63011. Remember or jot down the server name.

    4. Click on Portal and then Server Settings.

    5. Click on the server from step #4 (in this case \"node630011\")

    6. In the field IAX Port change the port number to the port configured in \"Changing iax.conf\" above. In this example, the port will be 4570.

    7. Click Submit

    Propagation of the change takes 1-2 minutes.

    "},{"location":"adv-topics/multinodesnetwork/#telling-local-nodes-about-their-neighbors","title":"Telling Local Nodes About Their Neighbors","text":"

    Each node on the same network needs to be hardcoded with information about the other node. In this example, node 630010 needs to be hardcoded for where 630011 is and vice-versa.

    "},{"location":"adv-topics/multinodesnetwork/#configuring-node-630010","title":"Configuring Node 630010","text":"

    As root (i.e. sudo -s) edit the file /etc/asterisk/rpt.conf. Locate the line with the node's own definition. For example:

    630010 = radio@127.0.0.1/630010,NONE\n

    After this line, add its local neighbor by its IP address and the alternative IAX port. In this example, add 630011 as follows:

    630010 = radio@127.0.0.1/630010,NONE\n630011 = radio@192.168.0.11:4570/630011,NONE\n

    Note the inclusion of the port as :4570. This pattern can be repeated for multiple nodes on the same network. Save the file and restart asterisk with systemctl restart asterisk.

    "},{"location":"adv-topics/multinodesnetwork/#configuring-node-630011","title":"Configuring Node 630011","text":"

    The other node repeats same process but in the opposite direction. As root (i.e. sudo -s) edit the file /etc/asterisk/rpt.conf. Locate the line with the node's own definition. For example:

    630011 = radio@127.0.0.1/630011,NONE\n

    After this line, add its local neighbor by its IP address and the standard IAX port since it is unchanged. In this example, add 630010 as follows:

    630011 = radio@127.0.0.1/630011,NONE\n630010 = radio@192.168.0.10:4569/630010,NONE\n

    Note the inclusion of the port as :4569. This pattern can be repeated for multiple nodes on the same network. Save the file and restart asterisk with systemctl restart asterisk.

    "},{"location":"adv-topics/multinodesnetwork/#testing-linking-locally","title":"Testing Linking Locally","text":"

    Test linking the two nodes together to ensure configuration. In this example, node 630010 will connect to node 630011 using the Asterisk CLI:

    rpt cmd 630010 ilink 3 630011\n

    Then the link should be shown to 630011 with the command:

    rpt nodes 630010\n

    that should display:

    node630010*CLI> rpt nodes 630010\n\n************************* CONNECTED NODES *************************\n\nT630011\n

    Disconnect the links with:

    rpt cmd 630010 ilink 1 630011\n

    After that, everything should work. The most common issue with this configuration is incorrect router/NAT port forwarding or a typo in rpt.conf.

    "},{"location":"adv-topics/noderesolution/","title":"Node Resolution","text":"

    app_rpt has three ways to resolve a node number to an IP address.

    The above mechanisms are used to translate a node number into its respective public IP address. Connections are made directly between nodes using their public IP addresses.

    app_rpt first looks to see if the node is statically configured. If it is not found, the setting for node_lookup_method in the [general] stanza of rpt.conf determines the next step. The default setting is to perform a DNS lookup and if not found, attempt a lookup using the external file.

    The operation of this ASL3 feature is controlled by the following information in /etc/asterisk/rpt.conf.

    [general]\nnode_lookup_method = both   ;method used to lookup nodes\n                    ;both = dns lookup first, followed by external file (default)\n                    ;dns = dns lookup only\n                    ;file = external file lookup only\n
    The node lookup routines will output debug information showing the node lookups if the debug level is set to 4 or higher.

    Note: The AllStarLink network maintains security by requiring node owners to register with AllStarLink. Before a node is published to our directory it must successfully authenticate and register with our server. Related see HTTP Registration

    "},{"location":"adv-topics/noderesolution/#external-node-directory-file","title":"External Node Directory File","text":"

    app_rpt can use an external node directory file to perform node resolution. The asl3-update-nodeList service is used to periodically download the current list of authenticated nodes.

    The name of this file is configured in rpt.conf using the extnodefile key and value. By default the name of this file is /var/lib/asterisk/rpt_extnodes. Multiple files can be specified based on your requirements. The file names are separated with a comma.

    When a file based lookup is performed, app_rpt reloads the external file before searching for a match. This allows external scripts the opportunity to keep the file updated.

    Here is an example of the external file format:

    [extnodes]\n\n2000=radio@162.248.93.134:4569/2000,162.248.93.134\n2001=radio@162.248.93.134:4569/2001,162.248.93.134\n
    "},{"location":"adv-topics/noderesolution/#domain-name-service-dns","title":"Domain Name Service DNS","text":"

    AllStarLink now provides a DNS service for performing node resolution. DNS is an Internet standard and is more efficient than using a file base solution.

    When using DNS app_rpt must know the length of the longest node number. The current maximum is 6 digits. This number can be overridden in rpt.conf by adding the max_dns_node_length key and value to the [general] stanza.

    For more information on DNS see AllStarLink DNS Servers.

    "},{"location":"adv-topics/noderesolution/#statically-configured-node","title":"Statically Configured Node","text":"

    In some cases, it is desirable to statically define the node information. This is commonly used for private nodes or private networks.

    To configure a static node, enter the information for the node in rpt.conf under the [node] stanza. Here is an example configuration:

    [nodes]\n1001 = radio@192.168.0.1/1001,NONE\n1998 = radio@127.0.0.1:4568/1998,NONE\n1999 = radio@192.168.0.2:4569/1999,NONE\n
    "},{"location":"adv-topics/other-software/","title":"Other Software Products","text":"

    The following information relates to software components related to AllStarLink.

    "},{"location":"adv-topics/other-software/#web-dashboards","title":"Web Dashboards","text":"

    The primary web dashboard supported by AllStarLink is Allmon3.

    But, we know there are other dashboards available and everyone has their own personal favorite. We also know that some of the other dashboards make use of database (\"astdb.txt\"). Typically, this DB has been downloaded by a PHP script (\"astdb.php\") and updated with the help of boot scripts and cron jobs.

    To support these other dashboards we have included a systemd service to download and update the \"astdb.txt\" database. By default, this service and its associated timer are disabled. If you have installed a dashboard that is dependent on this database you will need to execute the following commands :

    systemctl enable asl3-update-astdb.service\nsystemctl enable asl3-update-astdb.timer\nsystemctl start asl3-update-astdb.timer\n
    "},{"location":"adv-topics/permissions/","title":"Permissions","text":""},{"location":"adv-topics/permissions/#asterisk-vs-root","title":"\"asterisk\" vs \"root\"","text":"

    One of the new features with ASL3 is that the Asterisk process now runs as the \"asterisk\" user (not as the super-user; \"root\"). While sometimes handy this was potentially very dangerous. This change was made for increased security. It's a good thing!

    We acknowledge that there are some applications, commands, and scripts that you may want to bring over from older (and other) versions of ASL and that these may be impacted by ASL3's heightened security. Porting these over to ASL3 may require some simple changes to filesystem permissions. Other issues may need other solutions (e.g. you can't execute a command that requires \"root\" privileges when you are not the \"root\" user). For guidance, reach out to the AllStarLink Community.

    In short, in ASL3 the Asterisk process now runs as the \"asterisk\" user. If you modify the \"/etc/asterisk/rpt.conf\" file to execute a command (or script) it will NOT have super-user (\"root\") privileges. The command itself and any file it references must be readable by the \"asterisk\" user. Any files that the command needs to modify must be writable by the \"asterisk\" user. Lastly, the parent directory of any file being created must also be writable.

    Asterisk User and Sudo

    Do not give general access to sudo to the asterisk user. This is very dangerous and poor practice. While people will state \"I've always done it this way\" it is, in fact, a serious security risk. As AllStarLink is run on the public internet for many repeater sites, the development team prioritizes security of the system. For common uses of this, namely restarting services and shutting down a node, see Managing Services and OS Shutdowns/Reboots.

    "},{"location":"adv-topics/permissions/#filesystem-permissions","title":"Filesystem Permissions","text":"

    The following is a very brief overview of Linux filesystem permissions. More in-depth information is available on the internet (search for \u201clinux file permissions\u201d).

    On a Linux system, each file (and directory) has permissions that effect access by the \u201cowner\u201d, permissions that effect access to those logins in the same \u201cgroup\u201d, and permissions that effect access to \u201cother\u201d logins. The permissions determine whether you (or some other process) can read, write, or execute the files (or search in directories). There are commands to change the ownership (\u201cchown\u201d) and permissions (\u201cchmod\u201d) for each file/directory.

    "},{"location":"adv-topics/permissions/#file-permissions","title":"File permissions","text":"

    You can use the ls -l [FILE] command to view file permissions. The permissions determine whether the calling process can read (\"r\"), write (\"w\"), or execute (\"x\") the file. Here are some examples :

    node63001:~/examples $ ls -l *file*\n-rw------- 1 root root     0 Aug  5 08:28 1-file-owner-root-can-read-write\n-rw-r----- 1 root asterisk 0 Aug  5 08:28 2-file-add-group-asterisk-can-read\n-rw-r--r-- 1 root asterisk 0 Aug  5 08:27 3-file-add-everyone-can-read\n\nnode63001:~/examples $ ls -l *script*\n-rwx------ 1 root root     0 Aug  5 08:28 1-script-owner-root-can-read-write-execute\n-rwxr-x--- 1 root asterisk 0 Aug  5 08:28 2-script-add-group-asterisk-can-read-execute\n-rwxr-xr-x 1 root asterisk 0 Aug  5 08:27 3-script-add-everyone-can-read-execute\n

    In each case, the file permissions progress from the most restrictive (only the file owner can read/execute) to most available (all users can read/execute).

    "},{"location":"adv-topics/permissions/#directory-permissions","title":"Directory permissions","text":"

    You can use the ls -ld <directory> command to view the permissions on a directory (vs. the contents of a directory). The permissions determine whether the calling process can read (\"r\"), write (\"w\"), or search (\"x\") the directory. Here are some examples :

    node63001:~/examples $ ls -ld *dir*\ndrwx------ 2 root root     4096 Aug  5 09:56 1-dir-owner-root-can-read-write-search\ndrwxr-x--- 2 root asterisk 4096 Aug  5 09:56 2-dir-add-group-asterisk-can-read-search\ndrwxr-xr-x 2 root asterisk 4096 Aug  5 09:56 3-dir-add-everyone-can-read-search\n

    In each case, the directory permissions progress from the most restrictive (only the directory owner can read/write/search) to most available (all users can read/search).

    "},{"location":"adv-topics/permissions/#changing-permissions","title":"Changing permissions","text":"

    The following commands can be used to change the permissions of files and directories :

    Command Description Sample Usage chown change file owner and group # chown root /var/asl-backups/asl-backup-files chgrp change group ownership # chgrp asterisk /var/spool/asterisk/monitor chmod change file mode bits # chmod 640 /etc/asterisk/manager.conf

    Any process with \"write\" permission to a file can update (including completely overwrite) the file. A process with \"write\" permission to a directory can add/remove/rename files in that directory. If a command/script is writable then it can be changed to \"do something different\" when it is next executed. When updating filesystem permissions, please remember that doing so can potentially expose the contents of files (and directories) to others. Be cautious!

    "},{"location":"adv-topics/permissions/#managing-services-and-os-shutdownsreboots","title":"Managing Services and OS Shutdowns/Reboots","text":"

    Beginning with asl3-3.4.0, AllStarLink v3 comes with a PolicyKit ruleset to permit the asterisk user to execute a limited number of actions without the need for sudo or prompting for a password. Those are:

    Using a combination of wrapper scripts and appropriate function configuration, Asterisk can restart itself, Allmon3, shutdown the system, or reboot the system. Use of the wrapper scripts for systemctl commands is essential for a clean execution of the scripts under the polkit rules. The provided wrappers are:

    These privileges can be used inside /etc/asterisk/rpt.conf within the [functions] stanza as follows:

    9001 = cmd,/etc/asterisk/scripts/asterisk-restart\n9002 = cmd,/etc/asterisk/scripts/asterisk-stop\n9003 = cmd,/etc/asterisk/scripts/allmon3-restart\n9004 = cmd,/etc/asterisk/scripts/allmon3-stop\n9005 = cmd,/usr/sbin/shutdown\n9006 = cmd,/usr/sbin/reboot\n

    The command *9001 would restart asterisk, *9006 would reboot the system, etc.

    "},{"location":"adv-topics/sa818modules/","title":"SA818 RF Modules","text":"

    Setting up an SA818 RF module is much easier on ASL3.

    Two new commands were added to ASL3 Pi Appliance systems to program the module. We believe that most will prefer using sa818-menu, a visual interface to the RF module settings. But, for those who prefer a command line interface we have also included sa818.

    "},{"location":"adv-topics/sa818modules/#sa818-menu","title":"sa818-menu","text":"

    To program your SA818 with the menu executing the following command from the Linux shell :

    sa818-menu\n

    Here, you will be able to view the last saved (or default) settings. Please note that the \"last saved\" settings are those saved by the sa818-menu command on your system. We have no way to \"read\" settings from the SA818 RF module.

    The settings include :

    Setting Description Band Specify the module type (VHF, UHF) Bandwidth Narrow (12.5KHz) or Wide (25.0KHz) Receive Frequency Transmit Frequency Squelch Value Default = 1 Volume Default = 1 Sub-audible tone None, CTCSS, DCS Pre-Emphasis/De-emphasis Default = Disabled High pass Filter Default = Disabled Low pass Filter Default = Disabled Serial Port Default = /dev/serial0 or /dev/ttyUSB0 Connection Speed Default = 9600

    If you change the sub-audible tone setting to CTCSS or DCS you will be presented with a list of possible tone frequencies. When using CTCSS tones, you will also have a setting for the CTCSS Reverse Burst (Tail Tone).

    The RF module will be programmed when you exit the menu.

    Note: if you make changes that you do not wish to have stored then use the ESC key to exit the menu.

    "},{"location":"adv-topics/sa818modules/#sa818","title":"sa818","text":"

    To program your SA818 using the command line you can execute the sa818 command. For command usage information, use sa818 --help.

    "},{"location":"adv-topics/sip-phone/","title":"Setting up a SIP Phone","text":"

    This document describes the steps necessary to set up a SIP phone in ASL3. The setup procedure has changed due to the deprecation of chan_sip. Users are now required to use chan_pjsip. chan_pjsip brings new features to the SIP stack and is the supported SIP channel for the future.

    "},{"location":"adv-topics/sip-phone/#update-modulesconf","title":"Update modules.conf","text":"

    chan_pjsip requires a number of modules to be loaded. You should start by editing /etc/asterisk/modules.conf and add the following to the [modules] stanza (generally near the bottom of the file just above the [global] stanza):

    ;\n; modules for pjsip\n;\nnoload = app_voicemail.so\nload = bridge_builtin_features.so\nload = bridge_builtin_interval_features.so\nload = bridge_holding.so\nload = bridge_native_rtp.so\nload = bridge_simple.so\nload = bridge_softmix.so\nload = chan_bridge_media.so\nload = chan_pjsip.so\nload = func_pjsip_endpoint.so\nload = func_sorcery.so\nload = func_devstate.so\nload = res_pjproject.so\nload = res_pjsip_acl.so\nload = res_pjsip_authenticator_digest.so\nload = res_pjsip_caller_id.so\nload = res_pjsip_dialog_info_body_generator.so\nload = res_pjsip_diversion.so\nload = res_pjsip_dtmf_info.so\nload = res_pjsip_endpoint_identifier_anonymous.so\nload = res_pjsip_endpoint_identifier_ip.so\nload = res_pjsip_endpoint_identifier_user.so\nload = res_pjsip_exten_state.so\nload = res_pjsip_header_funcs.so\nload = res_pjsip_logger.so\nload = res_pjsip_messaging.so\nload = res_pjsip_mwi_body_generator.so\nload = res_pjsip_mwi.so\nload = res_pjsip_nat.so\nload = res_pjsip_notify.so\nload = res_pjsip_one_touch_record_info.so\nload = res_pjsip_outbound_authenticator_digest.so\nload = res_pjsip_outbound_publish.so\nload = res_pjsip_outbound_registration.so\nload = res_pjsip_path.so\nload = res_pjsip_pidf_body_generator.so\nload = res_pjsip_pidf_digium_body_supplement.so\nload = res_pjsip_pidf_eyebeam_body_supplement.so\nload = res_pjsip_publish_asterisk.so\nload = res_pjsip_pubsub.so\nload = res_pjsip_refer.so\nload = res_pjsip_registrar.so\nload = res_pjsip_rfc3326.so\nload = res_pjsip_sdp_rtp.so\nload = res_pjsip_send_to_voicemail.so\nload = res_pjsip_session.so\nload = res_pjsip.so\nnoload = res_pjsip_t38.so\nnoload = res_pjsip_transport_websocket.so\nload = res_pjsip_xpidf_body_generator.so\nload = res_rtp_asterisk.so\nload = res_sorcery_astdb.so\nload = res_sorcery_config.so\nload = res_sorcery_memory.so\nload = res_sorcery_realtime.so\n
    "},{"location":"adv-topics/sip-phone/#update-extensionsconf","title":"Update extensions.conf","text":"

    The next step is to update extension.conf. This creates a dial plan that controls access to your node by SIP phones. Edit /etc/asterisk/extensions.conf and add the following to the bottom of the file. This example will set up extension 1001 as your local phone and the dialing plan for your node.

    [sip-phones]\nexten => 1001,1,Dial(PJSIP/${EXTEN},60,rT)\n\nexten => ${NODE},1,Ringing\nexten => ${NODE},n,Answer(3000)\nexten => ${NODE},n,Set(NODENUM=${CALLERID(number)})\nexten => ${NODE},n,Playback(extension)\nexten => ${NODE},n,SayDigits(${NODENUM})\nexten => ${NODE},n,Playback(rpt/connected)\nexten => ${NODE},n,Playback(rpt/node)\nexten => ${NODE},n,SayDigits(${EXTEN})\nexten => ${NODE},n,rpt(${EXTEN}|P)\nexten => ${NODE},n,Hangup\n

    The above dial plan will allow your SIP phone to dial your node number. In this example one should use *99 to enable PTT and # to disable PTT from the SIP phone. VOX is also an option for phone connections. When using VOX, one may want to mute the mic on the phone when not transmitting and unmute to transmit. VOX is enable by adding a 'v' to the rpt app line like this:

    exten => ${NODE},n,rpt(${EXTEN}|Pv)\n

    This dial plan announces the extension number and the connecting node number. The lines with SayDigits and Playback can be removed if you do not want the announcement.

    The ${NODE} variable is defined at the top of extensions.conf and should be your assigned node number. See the following.

    [globals]\nHOMENPA = 999 ; change this to your Area Code\nNODE = 1999   ; change this to your node number\n

    The entry 1001 in extensions.conf is the extension number for your SIP phone. If you want to use a different extension number, change 1001 to your desired extension number.

    Note: If you attempt to copy extensions.conf from a previous release of AllStarLink, it will fail. AllStarLink version 3 requires comma delimiters instead of pipe symbols for standard asterisk functions. The rpt function continues to use a pipe delimiter. This is subject to change in a future release.

    "},{"location":"adv-topics/sip-phone/#update-pjsipconf","title":"Update pjsip.conf","text":"

    You are now ready to configure pjsip. Edit /etc/asterisk/pjsip.conf. Don\u2019t get overwhelmed by all of the text in the sample configuration. pjsip supports several configurations. This document focuses on the necessary entries to get a SIP phone operational.

    Scroll down to the Basic UDP transport section. It should be updated to look like the following:

    ; Basic UDP transport\n;\n[transport-udp]\ntype=transport\nprotocol=udp    ;udp,tcp,tls,ws,wss,flow\nbind=0.0.0.0:5060\n

    Note: After the bind port address (0.0.0.0) is the incoming port number (5060). This UDP port must not be blocked by your network firewall.

    Scroll down to the section titled Endpoint Configured For Use With A Sip Phone. For each extension, you will need to enter 3 sections. Here is an example for extension 1001.

    [1001]\ntype=endpoint\ntransport=transport-udp\ncontext=sip-phones\ndisallow=all\nallow=ulaw\nallow=alaw\nallow=gsm\nauth=1001\naors=1001\ncallerid=\"My CallerID\"\n;\n; A few more transports to pick from, and some related options below them.\n;\n;transport=transport-tls\n;media_encryption=sdes\n;transport=transport-udp-ipv6\n;transport=transport-udp-nat\n;direct_media=no\n;\n; MWI related options\n\n;aggregate_mwi=yes\n;mailboxes=6001@default,7001@default\n;mwi_from_user=6001\n;\n; Extension and Device state options\n;\n;device_state_busy_at=1\n;allow_subscribe=yes\n;sub_min_expiry=30\n;\n; STIR/SHAKEN support.\n;\n;stir_shaken=no\n;stir_shaken_profile=my_profile\n\n[1001]\ntype=auth\nauth_type=userpass\npassword=1001\nusername=1001\n\n[1001]\ntype=aor\nmax_contacts=4\n;contact=sip:6001@192.0.2.1:5060\n

    If you want to use a different extension number, you will need to change [1001] with your desired number. The item callerid=\"My CallerID\" should also be updated for your preferred caller identifier.

    The items password=1001 and username=1001 should be updated to match the username and password you want for your extension number. You should use a complex password. Do not use the extension number. You can use the password generator located here to generate a complex password.

    These three sections can be replicated for each extension that you want to add to the system.

    "},{"location":"adv-topics/sip-phone/#restart-asterisk","title":"Restart asterisk","text":"

    After completing these changes, you must restart asterisk or reboot your system.

    systemctl restart asterisk\n
    "},{"location":"adv-topics/sip-phone/#example-config-with-multiple-sip-phones","title":"Example config with multiple SIP phones","text":"

    As noted above you can have more SIP phones. Here is an example of a simple dial plan that has multiple extensions.

    [sip-phones]\n; Extension 210 - Line 1\n; Extension 211 - Line 2\n; Extension 212 - Garage\n; Extension 213 - Cordless ATA\n; Extension 1000 - Voice Mail\n\nexten => 210,1,Dial(PJSIP/210,60,rT)\nexten => 211,1,Dial(PJSIP/211,60,rT)\nexten => 212,1,Dial(PJSIP/212,60,rT)\nexten => 213,1,Dial(PJSIP/213,60,rT)\nexten => 1000,1,VoiceMailMain(210)\nexten => 1000,2,Hangup\n\n; Allow SIP calls to local nodes\nexten => 1998,1,rpt(1998|P)\nexten => 1999,1,rpt(1999|P)\n

    This example includes voicemail. To use voice mail, you will need to change noload = app_voicemail.soto load = app_voicemail.so.

    "},{"location":"adv-topics/sip-phone/#troubleshooting","title":"Troubleshooting","text":"

    If you have trouble connecting your SIP phone, start the asterisk command line with asterisk -rvvv. Enter the command pjsip set logger on. This will show the communications between the SIP phone and asterisk. It will also show the actions in your dial plan.

    "},{"location":"adv-topics/sip-phone/#security","title":"Security","text":"

    If you will be exposing your system to the outside world. You should consider using fail2ban to protect the system.

    "},{"location":"adv-topics/tts/","title":"ASL Text-to-Speech","text":"

    ASL provides the asl-tts command. This is a text to speech speaker for ASL3. It uses the piper open source TTS system that is not depending on cloud providers and has opensource voice models for dozens of languages.

    "},{"location":"adv-topics/tts/#basic-operation","title":"Basic Operation","text":"

    asl-tts uses the piper-tts text to speech engine to generate ulaw files from text so that Asterisk/apt_rpt can speak any set of arbitrary text without needing to have a sound file installed for the word necessary. By default, asl-tts will cause app_rpt to immediately speak whatever text was specified.

    asl-tts has a time delay

    Running the piper-tts system takes a few seconds to compile the voice into the sound file. Larger text blocks may take tens of seconds to compile. asl-tts is not appropriate for time-sensitive communications such as speaking the current time.

    The command format is as follows: 

    asl-tts -n NODE -t 'TEXT' [ -v VOICE ] [ -f FILE ]\n

    The general use case is to speak to a node immediately. This this case speaking the text \"Good morning\" to node 63001 would look like this:

    sudo asl-tts -n 63001 -t 'Good morning'\n

    The TTS engine will start, create the audio, and execute the playback on the node. The playback follows normal app_rpt rules and will be played back as soon as its turn arrives. The command will not preempt or \"talk over\" other traffic on the node.

    Keep in mind the differences between quoting in a shell command. In general, the \" and ' work the same by the double-quote will allow variable substitution while the single quote will not. This can be useful if you want to create text based on a system condition. For example:

    TODAY=$(date +\"%A, %B %d, %Y\")\nsudo asl-tts -n 63001 -t \"Today is ${TODAY}\"\n

    Obviously the above example is a bit contrived as the date command doesn't generate nice words such as \"first\", \"seventeenth\", etc. However it demonstrates the flexibility of the tool. A script could be used to populate $TODAY with natural words. For a hypothetical example:

    TODAY=$(/path/to/today-script)\nsudo asl-tts -n 63001 -t \"Today is ${TODAY}\"\n

    More complex message can be created with additional shell coding. As an example here's how the current public IP address of the system could be obtained and spoken:

    IPADDR=$(wget -q -O - checkip.dyndns.com | grep -Po \"[\\d\\.]+\" | sed 's/\\./ dot /g')\nMSG=\"The node public IP is ${IPADDR}\"\nsudo asl-tts -n 63001 -t ${MSG}\n

    A bash shell variable can generally hold up to a 2 Mb-sized string so any reasonably-formatted text can be stored in a variable and then played with the -t command. Keep in mind that the text or the variable contents must not contain the same quote type used in the command without escaping it with a backslash. For example \"Nested \\\" mark\".

    Temporary File Cleanups

    Note that temporary files are written to directories named /tmp/asl-tts and are not automatically cleaned up because it's impossible to know when Asterisk will actually need/speak the file contents. If this becomes a space problem, put in a systemd timer unit or a cron job to delete old files in /tmp/asl-tts. The command can simply be rm -rf /tmp/asl-tts*.

    "},{"location":"adv-topics/tts/#creating-a-sound-file-for-later","title":"Creating a Sound File for Later","text":"

    To create the text for other usage, such as cron jobs, use the \"-f FILE\" option to specify where the file should be created. Do not use an extension; the file will be automatically appended with the .ul postfix.

    For example: 

    AUD_TEXT=\"The club net will begin in five minutes!\"\nasl-tts -n 460181 -t \"${AUD_TEXT}\" -f /tmp/netstartsin5\n

    This will create a sound file /tmp/netstartsin5.ul that can be used later with an asterisk playback such as: 

    asterisk -rx 'rpt playback 63001 /tmp/netstartsin5\n

    Note that -n is required but irrelevant to this use case.

    "},{"location":"adv-topics/tts/#voices","title":"Voices","text":"

    By default, the package provides on voice \"Amy\" from https://github.com/rhasspy/piper/blob/master/VOICES.md. This voice is the closest voice to the default Asterisk and app_rpt sounds. It is possible to download other voices, for any language, and store the .onnx and .onxx.json files in /var/lib/piper-tts and then use them with the -v option. For example, to add the English en_GB voice alan from the voices repository:

    1. Download the .onnx and .onnx.json file to /var/lib/piper-tts

    2. Specify -v en_GB-alan-low.onnx on the command line to asl-tts. As an example:

    asl-tts -n 63001 -v en_GB-alan-low.onnx -t \"Good morning\"\n

    Note, since all files are squashed down to 8K uLaw format, there is no value in the \"medium\" or \"high\" quality models. Always use the low quality model.

    "},{"location":"adv-topics/tts/#reporting-bugs","title":"Reporting Bugs","text":"

    Report bugs to https://github.com/AllStarLink/asl3-tts/issues

    "},{"location":"adv-topics/uefi-secureboot/","title":"UEFI and SecureBoot","text":"

    UEFI (Unified Extensible Firmware Interface) is a modern firmware interface that replaces the older BIOS system, providing a more flexible and powerful environment for booting and managing hardware. It supports larger hard drives, faster boot times, and a more user-friendly interface.

    Secure Boot is a feature of UEFI that ensures only trusted software is loaded during the boot process, protecting the system from malware and unauthorized software. It works by verifying the digital signatures of boot loaders and operating system files against a database of trusted certificates.

    Together, UEFI and Secure Boot enhance the security and efficiency of modern computing systems.

    Given a quirk of how Asterisk + app_rpt needs a kernel module (driver) not already contained in the core system provided by Debian, installing AllStarLink on a system requiring UEFI/SecureBoot requires additional steps to create what's called a Machine Owner Key.

    Note

    Raspberry Pi does not use or support UEFI or SecureBoot

    "},{"location":"adv-topics/uefi-secureboot/#generating-a-machine-owner-key","title":"Generating a Machine Owner Key","text":"

    Generation of a Machine Owner Key (MOK) is can be done with the asl-setup-dkms-mok utility. This tool appeared in the package asl3-3.3.0-1.

    1. Run the command sudo asl-setup-dkms-mok. This will prompt for several items of information. An example run will look like:

      $ sudo asl-setup-dkms-mok\n!!!\n!!! WARNING - RUNNING THIS SCRIPT WILL REGENERATE THE\n!!! DEFAULT MACHINE OWNER KEY (MOK) FOR SECURE BOOT. IN GENERAL YOU\n!!! ARE RUNNING THIS SCRIPT BECAUSE YOU NEED TO AND YOU KNOW WHY\n!!! YOU ARE DOING IT. HOWEVER BE WARNED!\n!!!\nENTER \"galaxy\" TO CONTINUE>> galaxy\n\nGenerating an MOK requires a name for the certificate. This can be\nanything, however your name or your callsign is recommended.\n\nName or callsign>> N8EI\n\nGenerating the MOK...\n......+...+...+.....+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*................+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*.+.........+......+...+.+...........+...+.+...+..+.+...+..............+.......+...........+.........+................+..+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++\n.....+............+.........+.+.....+....+...+...+..............+.+..+...............+.+..+...+.+........+.+......+........+...............+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*.......+...+............+..+....+............+..+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*....+.....+...+...........................+......+.......+...............+.....+......+..........+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++\n-----\n\nEnter a one-time password for this key. This password will be prompted\nfor at the next reboot to properly install the key in the UEFI\nbootloader keyring. This does not need to be an incredibly complex\npassword and you may write it down until it's used.\n\ninput password:\ninput password again:\n\nSetup /etc/dkms/framework.conf.d/asl.conf\nGenerate /etc/dkms/asl_sign_helper.sh\n\nYou must reboot the system and enter the OTP to load the key into SecureBoot\nThis will appear on your directly-connected monitor or your cloud console\n

    2. Prepare to Reboot the System. Make sure a monitor is attached to the system or, if using a cloud virtual machine / VPS, make sure that a remote virtual console is connected. Then, reboot the system.

      MOK Load Staging is One-Time Only

      Note that the MOK is only staged for load one time. If the MOK load prompt is not responded to at the first/next reboot, the MOK staged for load is wiped and Step #1 will have to be run again.

    3. Complete the steps to review, unlock, and load the MOK

      The next step varies by vendor

      The next step uses screenshots from a Linux/QEMU virtual environment. Every device manufacturer or cloud provider will have a slightly different process or look.

    4. Reboot the system back into Linux and log in.

    5. Check that the MOK created is now loaded by looking for the \"name\" you provided the certificate in step 1. In this example, the MOK was named \"N8EI\".

      sudo dmesg | grep -i N8EI\n

      The output will look like: 

      $ sudo dmesg | grep -i N8EI\n[    0.674788] integrity: Loaded X.509 cert 'N8EI: 92ed492eeb11594eb25af5cc70ff0587115f2577'\n

    6. Rebuild the dahdi-dkms package so that the modules are signed by the MOK so that the kernel will load then.

      apt install --reinstall -y dahdi-dkms\n

    7. Test the module loads successfully:

       $ sudo modprobe dahdi dahd_dummy dahdi_transcode\n$ lsmod | grep dahdi\ndahdi_dummy            16384  0\ndahdi_transcode        16384  0\ndahdi                 258048  2 dahdi_dummy,dahdi_transcode\n
    "},{"location":"adv-topics/usbinterfaces/","title":"USB Audio Interfaces","text":"

    Setting up USB audio interfaces is much easier with ASL3.

    The ASL3 menu and Asterisk CLI USB config commands handle these changes.

    "},{"location":"adv-topics/usbinterfaces/#usb-interface-ic-support","title":"USB Interface IC Support","text":"

    Nearly all USB Interface products that are used with ASL nodes use C-Media CM108 and CM119 USB Interface ICs. These ICs provide the following features:

    The CM119 provides an additional 4 GPIO lines (GPIOs 5-8) vs. the CM108. The original CM108 and CM119 ICs were replaced in the early 2010's with the CM108AH and CM119A. These ICs are still used in many USB Radio Interfaces (URIs). Newer versions of the CM1xx ICs \u2013 the CM108B and CM119B are now available and have some minor differences in gain settings and other specifications but are fully compatible with the earlier IC versions and with ASL.

    "},{"location":"adv-topics/usbinterfaces/#setting-audio-levels","title":"Setting Audio Levels","text":"

    When setting up a new node, or if any changes have been made \u2013 such as use of a different radio, microphone, or other hardware or gain settings \u2013 checking and setting audio levels is very important. Radio equipment such as HTs or mobile radios, and many electronics products such as VOIP phones often restrict end-user audio adjustment capabilities to ensure consistent audio levels. Allstar nodes and USB interfaces however generally allow greater flexibility and thus extra attention is required to be sure your audio interface gains are properly matched to the external audio input and output levels. Otherwise your audio levels could be much quieter or louder than they should be which could be disruptive to other users and systems.

    Most nodes use the SimpleUSB channel driver. Nodes that require more advanced features such as DSP squelch detection, CTCSS tone encode/decode or additional audio filtering options will use the USBRadio channel driver. These channel drivers both have tune utilities for setting audio levels and other functions. Once a node has been configured in the asl-menu \"AllStar Node Setup Menu\", select the \"Interface Tune CLI\" menu option. The following options should then be used to check and optimize audio level settings:

    Note: Be sure you are not connected to any other nodes when doing any of the following tests and adjustments.

    The tune utilities can also be run from the command line without needing to use asl-menu. For SimpleUSB run \"sudo /usr/sbin/simpleusb-tune-menu\" or for USBRadio run \"sudo /usr/sbin/radio-tune-menu\".

    "},{"location":"adv-topics/usbinterfaces/#eeprom-operation","title":"EEPROM Operation","text":"

    The SimpleUSB and USBRadio channel drivers allow users to store configuration information in a EEPROM IC attached to the C-Media CM1xx USB Interface IC. The CM119A can have manufacturer information in the same area that stores the user configuration. The CM119B does have manufacturer data in the area that stores user configuration. The manufacturer data cannot be overwritten. The user configuration data has been moved higher in memory to prevent overwriting the manufacturer data. If you use the EEPROM to store configuration data, you'll need to save it to the EEPROM after upgrading. Use susb tune save or radio tune save.

    "},{"location":"allmon3/basics/","title":"Basics","text":"

    Allmon is the standard web-based monitoring and management for the AllStarLink v3. Allmon3 is the next generation of web dashboards for AllStarLink, rewritten for performance, scalability, and responsiveness.

    "},{"location":"allmon3/basics/#design-goals","title":"Design Goals","text":"

    Allmon3 features and functionality shall be governed by the following guidelines:

    "},{"location":"allmon3/basics/#community","title":"Community","text":"

    All code of Allmon3 not otherwise licensed (e.g., Bootstrap, etc.) is licensed under the GNU Affero General Public License v3. The choice of the AGPLv3 promotes giving back to the amateur radio and ASL communities.

    "},{"location":"allmon3/config/","title":"Configuration","text":"

    All configuration resides in /etc/allmon3. The stock configuration files are always available at /usr/share/doc/allmon3/ for recovery and documentation.

    After changing any configuration file, the service allmon3 must be restarted using systemctl restart allmon3.

    "},{"location":"allmon3/config/#node-configuration","title":"Node Configuration","text":"

    Allmon3's node connections are configured in /etc/allmon3/allmon3.ini. The allmon3.ini file is a standard INI-formatted file. Each stanza in the file is a node number. Each node stanza then can take the following options:

    Option Req'd / Opt Default Description host Req'd - DNS name or IP address of the Asterisk/ASL node port Opt 5038 Port of the Asterisk manager user Req'd - username of the Asterisk monitor, most commonly 'admin' pass Req'd - password of the monitor user multinodes Opt - this node is a server hosting multiple nodes and this is the \"primary\" record for the host. voters Opt n[,n,...] List voters on this server, comma separate. No value disabled voters pollinterval Opt 1 polling interval to asterisk in default is 1. this value can be expressed as a decimal fraction of a second - e.g., .5 is 500ms, .375 is 375ms, etc. vpollinterval Opt 1 Broadcast interval of voter data in seconds default is 1. this value can be expressed as a decimal fraction of a second - e.g., .5 is 500ms, .375 is 375ms, etc. retryinterval Opt 15 Seconds between retries if initial connection to asterisk is lost (optional, default 15) retrycount Opt \"Infinite\" number of times to retry a lost asterisk connection before ending (default infinite)

    An example minimal node configuration would be: 

    [12345]\nhost = 127.0.0.1\nuser = admin\npass = Passw0rd\n

    An example of a server with multiple nodes on it: 

    [12345]\nhost = 127.0.0.1\nuser = admin\npass = Passw0rd\nmultinodes = 12345, 23456\n

    Here's an example for monitoring three ASL Nodes: 

    [50815]\nhost=172.17.16.36\nuser=admin\npass=password\n\n[460180]\nhost=172.17.16.217\nuser=admin\npass=password\n\n[48496]\nhost=192.2.0.145\nuser=admin\npass=password\nvoter=y\nvotertitle=Megavoter\n

    "},{"location":"allmon3/config/#server-customization","title":"Server Customization","text":"

    Allmon3 has multiple configuration files to consider:

    "},{"location":"allmon3/config/#webini","title":"web.ini","text":"

    The web.ini file has four configuration sections - web, syscmds, node-overrides, and voter-titles.

    "},{"location":"allmon3/config/#web","title":"[web]","text":"

    The web section has the basic customizations for the Allmon3 site. Each item in this section is documented in the file.

    "},{"location":"allmon3/config/#syscmds","title":"[syscmds]","text":"

    The syscmds section defines the templates in the \"system commands\" menu. This stanza lists commands that are templates for the systems command modal dialog. The format is:

    [syscmds]\n  command text = Command Label\n  command text = Command Label\n  command text = Command Label\n  ...\n
    In any command text the @ will be replaced with the node the command modal was selected from. For example: 
        rpt status @ = Show Node Status\n
    wll result in the command 'rpt status 1999' assuming this command was selected from Node #1999.

    It is possible to create optional stanzas named in the format [syscmds-NODE] (e.g. [syscmds-1999]) and have those commands templates appear only for the given NODE listed. Example:

    [syscmds-1999]\nrpt cmd @ cop 999 xxx = Execute function 999\n
    "},{"location":"allmon3/config/#node-overrides","title":"[node-overrides]","text":"

    The node-overrides section can be used to override information from the ASL database. Using this section, the AllStarLink DB labels for a node may be replaces with custom text. The format is:

    [node-overrides]\n    NODE1 = TEXT HERE\n    NODE2 = TEXT HERE 2\n
    This section must exist even if it's empty. For example:

    [node-overrides]\n    1999 = My special private node\n    2010 = Wild Party Node\n
    "},{"location":"allmon3/config/#voter-titles","title":"[voter-titles]","text":"

    The voter-titles section is used to set display names for voters. This functions identically to [node-overrides]. For any voter not named here, voters will have an auto-generated name of \"Voter NODE\".

    "},{"location":"allmon3/config/#menuini","title":"menu.ini","text":"

    If this file is present in the api/ subdirectory of Allmon3's web interface and named menu.ini, this file will override the default behavior of the web interface simply listing the configured nodes in the left navbar. If this file is not present, all nodes in allmon3.ini/allmon3.ini.php will display singly in configured order.

    NOTE: Not listing a node in the menu WILL NOT cause the node to go un-polled if it is already configured in allmon3.ini. This allows the system administrator to have \"hidden\" nodes. If you want to completely stop polling an Asterisk/ASL node, the node must be removed from allmon3.ini.

    The format for this file is as follows:

    [ TITLE ]                         :: The stanza header is the label\n                                  :: for this menu item\ntype  = ( menu | single )         :: If type = menu, the item is interpreted as a dropdown\n                                  :: menu with each LABEL item displaying. If the\n                                  :: type = single, then only a the first LABEL = TARGET\n                                  :: will be displayed as a non-dropdown. In the case of\n                                  :: type = single, the [ TITLE ] is ignored completely\n\nLABEL = TARGET                    :: Each [ TITLE ] displays one or more LABELS\nLABEL = TARGET                    :: with a link to TARGET. When TARGET is all numbers\nLABEL = TARGET                    :: then TARGET is assumed to be a ASL node and\n                                  :: the link will filter down to the TARGET node\n                                  :: specified. Any other pattern will be interpreted\n                                  :: as a full or partial URL/URI.\n                                  ::\n                                  :: Of special node, creating a menu item of\n                                  :: multiple nodes to display, TARGET and simply be\n                                  :: #NODE,NODE,NODE - e.g. #1999,1998,1997\n
    Note that the order of the nodes within a stanza is irrelevant. They will be displayed as sorted alphabetically ascending according to UTF-8 (e.g. 0-9, A-Z, +).

    An example custom menu may look like: 

    [ W8WKY ]\ntype = menu\nW8WKY = 43211\n48496 = 48496\n45839 = 45839\n\n[ N8XPK ]\ntype = menu\nN8XPK = 42993\n43118 = 43118\n47987 = 47987\n\n[ Test ]\ntype = single\nAllStarLink = https://www.allstarlink.org\n

    "},{"location":"allmon3/config/#using-nginx-instead-of-apache","title":"Using Nginx instead of Apache","text":"

    Nginx can be used instead of Apache. Instead of using the apache2 package, install nginx using the above directions. After configuring nginx, edit /etc/nginx/sites-available/default (or your preferred site configuration) and add an include directive within the appropriate server { } configuration block. For example:

    server {\n    listen 80 default_server;\n\n    [... other stuff ...]\n\n    include /etc/allmon3/nginx.conf;\n\n    [... other stuff ...]\n}\n
    "},{"location":"allmon3/install/","title":"Installation","text":""},{"location":"allmon3/install/#supported-platforms","title":"Supported Platforms","text":"

    Allmon3 is only supported on:*

    "},{"location":"allmon3/install/#asl3-packages-install","title":"ASL3 Packages Install","text":""},{"location":"allmon3/install/#debian-12","title":"Debian 12","text":""},{"location":"allmon3/install/#debian-11","title":"Debian 11","text":""},{"location":"allmon3/install/#debian-10","title":"Debian 10","text":"

    Note: Allmon v1.2.1 will be the last version for Debian 10. Please upgrade your AllStarLink system to AllStarLink v3 on Debian 12.

    "},{"location":"allmon3/install/#install-from-source","title":"Install from Source","text":"

    Installation from source is not supported in the general use case. However, make install should yield a working system.

    "},{"location":"allmon3/usermgmt/","title":"User Management","text":"

    Allmon3 supports a robust multi-user configuration that permits per-node granular access as desired and that abstracts Allmon3 from the Asterisk AMI Manager configuration.

    Usernames and passwords are stored in /etc/allmon3/users.

    The default-configured username and password combination isallmon3 / password.

    You must change this before exposing a host to the Internet

    "},{"location":"allmon3/usermgmt/#user-database","title":"User Database","text":"

    Allmon3's user database is managed by allmon3-passwd. Adding a new user or editing an existing user is the same command. If the user does not exist, it will be added. If the user does exist, the password will be updated. To add or edit a user's password: 

    $ allmon3-passwd allmon3\nEnter the password for allmon3: password\nConfirm the password for allmon3: password\n
    That's all there is to it. The /etc/allmon3/users file is readable to see that the Argon2 hash changed for the user.

    Deleting a user is simply adding the --delete flag to the command:

    $ allmon3-passwd --delete allmon3\n

    After changing a user password, the allmon3 damon must be reloaded with systemctl reload allmon3.

    "},{"location":"allmon3/usermgmt/#per-node-restrictions-for-users","title":"Per-Node Restrictions for Users","text":"

    Allmon3 implements a lightweight access control system to restrict commands from certain users to certain nodes. Restrictions are configured in /etc/allmon3/user-restrictions. Given that the average use case is all users have similar access, the access control is implemented in a named-restrictions model for least configuration complexity.

    The logic is as follows when checking the restricted access list:

    1. If the user is not listed in user-restrictions at all than the user is permitted commands on all configured nodes.

    2. If the user is listed in user-restrictions and is listed as restricted to the node being commanded, the user is permitted to issue the command.

    3. If the user is listed in user-restrictions but the node is not listed for that user, the user is prohibited from issuing the command.

    The format of the user-restrictions file is:

    user1 | NODE[,NODE,NODE...]\nuser2 | NODE\n
    Lines beginning with # are comments.

    "},{"location":"developers/aptly/","title":"AllStarLink Apt Repos","text":"

    AllStarLink's primary installation method is through the use of Debian packages and the Apt deployment system.

    "},{"location":"developers/aptly/#repository-structure","title":"Repository Structure","text":"

    The repository is located at https://repo.allstarlink.org/public and is managed using Aptly. The Aptly structure is a multi-local-repo system that is coalesced into a single public repository with multiple components per distribution.

    There are three main local repositories:

    Each repository contains one or more distributions. Currently the only distribution supported is bookworm for Debian 12.

    Each repository has a default component name that, when published together, form the basis of the labeling scheme of promotion of packages from devel -> beta -> prod. The component mapping is:

    All local repositories are merged and published into a single public-facing repository. This is usable by one of the following .list configurations in/etc/apt/sources.list.d:

    # Primary AllStarLink Repo for Production Packages\ndeb [signed-by=/etc/apt/keyrings/allstarlink.gpg] https://repo.allstarlink.org/public bookworm main\n\n# Include Beta Component for Testing\ndeb [signed-by=/etc/apt/keyrings/allstarlink.gpg]  https://repo.allstarlink.org/public bookworm main beta\n\n# Include Beta and Devel component for Development\ndeb [signed-by=/etc/apt/keyrings/allstarlink.gpg]  https://repo.allstarlink.org/public bookworm main beta devel\n

    All packages in the repository are signed by GPG and the key is provided with repository installation .deb file.

    "},{"location":"developers/aptly/#aptly-github-runner-integrations","title":"Aptly + GitHub Runner Integrations","text":"

    ASL3-related GitHub repositories are integrated with the Aptly repository and running the \"Make and Publish Pkgs\" Action from each project will result in .deb files being compiled and stored in Aptly. Currently it's possible to build files directly into the devel and beta components but not prod. Moving packages into prod currently requires administrator intervention. This will be changed at some future date.

    "},{"location":"developers/aptly/#aptly-cli","title":"Aptly CLI","text":"

    It is possible to run Aptly commands from the shell on repo.allstarlink.org in addition to using the web-based API.

    "},{"location":"developers/aptly/#setup","title":"Setup","text":"

    All commands to Aptly must be run as the aptly user. To do this, each account on the repository server should be member of the sudo and aptly UNIX groups.

    Before issuing any Aptly commands run the following: 

    alias aptly=\"sudo -u aptly -- aptly\"\n
    to ensure that all commands are run as the aptly user.

    "},{"location":"developers/aptly/#adding-a-file-to-a-local-repo","title":"Adding a File to a Local Repo","text":"

    Adding a file to the local repo is straight-forward: 

    aptly repo add REPO FILE-or-DIR\n

    For example, to add the latest Allmon3 to the production repository: 

    $ aptly repo add asl3-prod allmon3_1.2.1-2_all.deb\nLoading packages...\n[+] allmon3_1.2.1-2_all added\n

    "},{"location":"developers/aptly/#publishing-repositories","title":"Publishing Repositories","text":"

    Coalescing and publishing the public repository with all components is: 

    aptly publish update bookworm\n
    When other distributions are supported, replace \"bookworm\" with the appropriate distribution. Upon execution, a prompt will appear for the GPG signing key for the repository. Provide it.

    And example, successful publication looks like: 

    $ aptly publish update bookworm\nLoading packages...\nGenerating metadata files and linking package files...\nFinalizing metadata files...\nSigning file 'Release' with gpg, please enter your passphrase when prompted:\nClearsigning file 'Release' with gpg, please enter your passphrase when prompted:\nCleaning up prefix \".\" components beta, devel, main...\n\nPublish for local repo ./bookworm [all, amd64, arm64, armhf] publishes {beta: [asl3-beta]}, {devel: [asl3-devel]}, {main: [asl3-prod]} has been successfully updated.\n

    "},{"location":"developers/aptly/#searching-for-packages","title":"Searching for Packages","text":"

    Listing and searching for packages inside of a local repo is done with the aptly repo search command and uses the (Aptly Package Query Language)[https://www.aptly.info/doc/feature/query/] to filter/select packages.

    A simple aptly repo search REPO will show all packages. For example: 

    $ aptly repo search asl3-prod\nallmon3_1.2.1-2_all\nasl-apt-repos_1.3-1.deb12_all\nasl3_3.0.0-1.deb_all\n[...]\n

    "},{"location":"developers/aptly/#search-by-name","title":"Search by name","text":"
    $ aptly repo search asl3-prod 'Name (asl3)'\nasl3_3.0.0-1.deb_all\n
    "},{"location":"developers/aptly/#search-by-name-glob-version","title":"Search by name glob + version","text":"
    $ aptly repo search asl3-prod 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12)'\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\n
    "},{"location":"developers/aptly/#search-by-name-glob-version-range","title":"Search by name glob + version range","text":"
    $ aptly repo search asl3-prod 'Name (~ asl3-asterisk*), Version (>= 1.0)'\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\n
    "},{"location":"developers/aptly/#promoting-packages-to-prod","title":"Promoting Packages to Prod","text":"

    Using the hypothetical example of promoting asl3-asterisk packages from beta to production.

    (1) Test package selection using aptly repo search 

    $ $ aptly repo search asl3-beta 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12)'\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_arm64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64\nasl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64\n

    (2) Move the packages into the production repository using aptly repo move 

    $ aptly repo move asl3-beta asl3-prod 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12)'\nLoading packages...\n[o] asl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all moved\n[o] asl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-tests_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-modules_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-dbgsym_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n[o] asl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all moved\n[o] asl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-modules-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all moved\n[o] asl3-asterisk-tests-dbgsym_2:20.7.0+asl3-1.0-5.deb12_arm64 moved\n[o] asl3-asterisk_2:20.7.0+asl3-1.0-5.deb12_amd64 moved\n

    (3) Update the repository publication 

    $ aptly publish update bookworm\nLoading packages...\nGenerating metadata files and linking package files...\nFinalizing metadata files...\nSigning file 'Release' with gpg, please enter your passphrase when prompted:\nClearsigning file 'Release' with gpg, please enter your passphrase when prompted:\nCleaning up prefix \".\" components beta, devel, main...\n\nPublish for local repo ./bookworm [all, amd64, arm64, armhf] publishes {beta: [asl3-beta]}, {devel: [asl3-devel]}, {main: [asl3-prod]} has been successfully updated.\n

    "},{"location":"developers/aptly/#note-on-all-architecture-packages","title":"Note on 'all' Architecture Packages","text":"

    The architecture all denotes that the package contains no binary compiled code and runs on any architecture. This is meaningful when handling asl3-asterisk packages for multiple architectures. Within the same version and release only one copy of the all packages needs to and CAN exist in the repository. The GitHub Runner builder will delete the all packages for the arm64 build since the arm64 and amd64 builds should occur simultaneously.

    To delete an extra set of all packages that may have ended up in the repository: 

    $ aptly repo search asl3-beta 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12), Architecture (all)'\nasl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all\nasl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all\n\n$ aptly repo remove asl3-beta 'Name (~ asl3-asterisk*), Version (2:20.7.0+asl3-1.0-5.deb12), Architecture (all)'\nLoading packages...\n[-] asl3-asterisk-config_2:20.7.0+asl3-1.0-5.deb12_all removed\n[-] asl3-asterisk-doc_2:20.7.0+asl3-1.0-5.deb12_all removed\n[-] asl3-asterisk-dev_2:20.7.0+asl3-1.0-5.deb12_all removed\n

    "},{"location":"developers/package-builds/","title":"Building ASL3-Asterisk","text":""},{"location":"developers/package-builds/#building-deb-files","title":"Building .deb Files","text":"

    These directions outline how to create the .deb files via GitHub using the Action workflows and the ASL custom runner AMIs in AWS.

    Kicking off the build processes requires knowledge of all of the topics below before starting it.

    "},{"location":"developers/package-builds/#package-version-format","title":"Package version format","text":"

    All packages created by this repo have the following version format based on deb-version(7) manpage explanation.

    \n ${EPOCH}:${ASTERISK_VERSION}+asl3-${RPT_VERSION}-${PACKAGE_VERSION}\n\n|----+---|--------+--------------------------X-----+--------X\n     |            |                                |\n     |            +-- \"upstream-version\"           |\n     |                                             |\n     +-- \"epoch\"               \"debian-revision\" --+\n

    The values are defined as follows:

    A file generated from this repo using the versioning format above will be named, for example asl3-asterisk-20.8.1+asl3-3.0.0-1. Note that the epoch does not appear in the filename by debian convention.

    "},{"location":"developers/package-builds/#determining-asterisk-version","title":"Determining Asterisk Version","text":"

    Currently, this is structured to be built against Asterisk 20 Long Term Servicing (LTS). Do not use any version of Asterisk that is not the latest version of Asterisk 20 LTS. The version is listed at Download Asterisk. The version of Asterisk entered in the Actions launcher will cause the proper version of Asterisk 20 LTS to be downloaded and folded into the builder.

    "},{"location":"developers/package-builds/#locking-in-and-determining-the-app_rpt-version","title":"Locking In and Determining the app_rpt Version","text":"

    When a build of .deb files is initiated, the app_rpt repo should must be tagged prior to initiating the build process. The tag version should be in the format MAJOR.MINOR.PATCH as found in app_rpt.h.

    "},{"location":"developers/package-builds/#determining-the-package-version","title":"Determining the Package Version","text":"

    This should be a monotonically increasing integer starting with 1 reflecting a change in Asterisk Version + app_rpt. For example, if building Asterisk 20.8.1 with app_rpt v3.0.0, then the first build of a package should be 1, the second 2, and so on.

    However, if from the above example, app_rpt is now v3.1.0 or Asterisk becomes 20.9.0 then the package version should reset back to 1 and start over.

    "},{"location":"developers/package-builds/#create-a-github-release","title":"Create a GitHub Release","text":"

    Create a Release to store the .deb files in. The name of the release should follow the format:

    ASTERISK_VERSION_asl3-RPT_VERSION-PACKAGE_VERSION\n

    For the above example of Asterisk \"20.7.0\", app_rpt \"1.2\", and package version \"1\" the release tag and name would be: 

    20.7.0_asl3-1.2-1\n
    Note: GitHub acts oddly about + signs in the tags and release names so it's converted to an underscore.

    "},{"location":"developers/package-builds/#executing-the-github-action-to-build-packages","title":"Executing the GitHub Action to Build Packages","text":"

    The following steps build the Debian packages:

    Note: as noted above, you will choose the platform architecture when starting a workflow. All workflows will create the architecture specific packages. The \"amd64\" workflow also creates the architecture independent (\"all\") packages. What does this mean? If you only want \"amd64\" packages then you would only need to run the single workflow. But, if you want packages for an alternate architecture (e.g. \"arm64\") then will need to run a workflow for that architecture AND one for the \"amd64\" architecture.

    "},{"location":"developers/source-install/","title":"Source-Based Installation","text":"

    Source install is for developers or users who need to install AllStarLink 3 on unsupported hardware or operating systems. Installing ASL from source code is primarily for developers. Doing so will require you to download, compile, and install multiple projects. You will also need to be very comfortable using various development tools and the Linux CLI.

    The following instructions are for building ONLY Asterisk with ASL's app_rpt.

    "},{"location":"developers/source-install/#install-phreaknetsh-script","title":"Install phreaknet.sh script","text":"

    The phreaknet script compiles and patches Asterisk and DAHDI. You should install the script with the following commands :

    cd /usr/src\nwget https://docs.phreaknet.org/script/phreaknet.sh\nchmod +x phreaknet.sh\n./phreaknet.sh make\n

    Once installed, you can keep the script updated with :

    phreaknet update\n
    "},{"location":"developers/source-install/#install-asterisk-20-lts","title":"Install Asterisk 20 LTS","text":"

    Use -t or -b for developer mode. Both are optional.

    phreaknet install -d -b -v 20\n

    Asterisk should be running at this point but not app_rpt. Check the install with asterisk -r.

    "},{"location":"developers/source-install/#clone-asl3-repo","title":"Clone ASL3 repo","text":"
    cd /usr/src\ngit clone https://github.com/AllStarLink/app_rpt.git\n
    "},{"location":"developers/source-install/#install-asl3","title":"Install ASL3","text":"

    This script does a git pull of app_rpt and compiles the branch you are on.

    cd app_rpt\n./rpt_install.sh\n
    "},{"location":"developers/source-install/#install-asl3-configs","title":"Install ASL3 configs","text":"

    This adds ASL3 configs to the full set of Asterisk configuration files. ASL3 modules.conf limits what actually runs.

    cp /usr/src/app_rpt/configs/rpt/* /etc/asterisk\n

    After restarting Asterisk (or rebooting the system) you should now have a complete ASL3 install. You can use the following command as a quick check :

    asterisk -rx \"rpt localnodes\"\n
    You should see node 1999.

    "},{"location":"mans/asl-find-sound/","title":"asl-find-sound","text":""},{"location":"mans/asl-find-sound/#name","title":"NAME","text":"

    asl-find-sound - Display app_rpt-compatible sound devices

    "},{"location":"mans/asl-find-sound/#synopsis","title":"SYNOPSIS","text":"

    usage: asl-find-sound

    "},{"location":"mans/asl-find-sound/#description","title":"DESCRIPTION","text":"

    asl-find-sound - Display app_rpt-compatible sound devices

    "},{"location":"mans/asl-find-sound/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-find-sound/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-node-lookup/","title":"asl-node-lookup","text":""},{"location":"mans/asl-node-lookup/#name","title":"NAME","text":"

    asl-node-lookup - Display AllStarLink node information

    "},{"location":"mans/asl-node-lookup/#synopsis","title":"SYNOPSIS","text":"

    asl-node-lookup [ --help ] [ --verbose ] [ --ns <name-server> ] <node#>

    Required arguments :

    <node#> : ASL node number to query

    Optional arguments :

    --help : show help

    --verbose : report additional information including the DNS \"SOA\" and \"NS\" records

    --ns : issue DNS queries to the specified name server

    Note: this command will also report Asterisk/rpt node lookup results if exec'd as \"root\" (or \"asterisk\").

    "},{"location":"mans/asl-node-lookup/#description","title":"DESCRIPTION","text":"

    asl-node-lookup - Display AllStarLink node information

    "},{"location":"mans/asl-node-lookup/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-node-lookup/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Allan Nathanson and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-play-arn/","title":"asl-play-arn","text":""},{"location":"mans/asl-play-arn/#name","title":"NAME","text":"

    asl-play-arn - Play Amateur Radio Newsline

    "},{"location":"mans/asl-play-arn/#synopsis","title":"SYNOPSIS","text":"

    usage: `asl-play-arn [-h] --node [ --when WHEN ] [ --debug ]

    optional arguments:

    "},{"location":"mans/asl-play-arn/#description","title":"DESCRIPTION","text":"

    Basic use is either immediately from the command line:

    allstar-play-arn --node 1999\n

    Depending on the processing speed of the device and Internet connectivity, the start of playback may take a significant time. If the desire is for precision on the start time, use the --when command and execute allstar-play-arn a few minutes before the desired start time.

    The script is silent except on errors like all good Unix utilities. Some useful troubleshooting may be done with the --debug option if an error is not revealing.

    "},{"location":"mans/asl-play-arn/#sceheduling-the-playback","title":"Sceheduling the Playback","text":"

    The best way to schedule the playback is as follows as the root user.

    1. cp /usr/share/asl3/asl-play-arn.* /etc/systemd/system

    2. Edit the OnCalendar= entry of /etc/systemd/system/asl-play-arn.timer for the timing of the playback desired.

    3. Edit the value for \"NODE\" in /etc/systemd/system/asl-play-arn.service for the node to execute the playback upon.

    4. Enable the timer unit: 

      systemctl daemon-reload\nsystemctl enable asl-play-arn.timer\n

    "},{"location":"mans/asl-play-arn/#asteriskapp_rpt-configuration","title":"Asterisk/app_rpt Configuration","text":"

    Usually the timeout timer in app_apt will be too short to accomodate the playing of the full news file. The following commands can be added to /etc/asterisk/rpt.conf to enable and disable the TOT:

    907=cop,7    ; Time out timer enable\n908=cop,8    ; Time out timer disable\n

    These appear as commented-out options in the stock rpt.conf

    "},{"location":"mans/asl-play-arn/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-play-arn/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-say/","title":"asl-say","text":""},{"location":"mans/asl-say/#name","title":"NAME","text":"

    asl-say - Cause Asterisk to speak

    "},{"location":"mans/asl-say/#synopsis","title":"SYNOPSIS","text":"

    usage: asl-say -n NODE -w ( time | time24 | ip4 | ip6 )

    "},{"location":"mans/asl-say/#description","title":"DESCRIPTION","text":"

    asl-say will speak the one of the following things on the node specified with -n as directed by -w

    time - The current time

    time24 - The current time in 24-hour format

    ip4 - The first IPv4 address of the system

    ip6 - The first global-scope IPv6 of the system

    "},{"location":"mans/asl-say/#example","title":"EXAMPLE","text":"

    asl-day -n 1999 -w time will speak the current time on node 1999.

    "},{"location":"mans/asl-say/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-say/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"mans/asl-setup-dkms-mok/","title":"asl-setup-dkms-mok","text":""},{"location":"mans/asl-setup-dkms-mok/#name","title":"NAME","text":"

    asl-setup-dkms-mok - Configure the Machine Owner Key (MOK) for signing kernel packages, notably dahdi* for ASL.

    "},{"location":"mans/asl-setup-dkms-mok/#synopsis","title":"SYNOPSIS","text":"

    usage: asl-setup-dkms-mok

    "},{"location":"mans/asl-setup-dkms-mok/#description","title":"DESCRIPTION","text":"

    This script is a walkthrough script for generating a UEFI Machine Owner Key (MOK). A MOK is needed on systems with UEFI SecureBoot to properly authenticate the DAHDI-related kernel modules which are rebuilt on kernel upgrades with DKMS. See the AllStarLink manual for more information or https://wiki.debian.org/SecureBoot.

    "},{"location":"mans/asl-setup-dkms-mok/#bugs","title":"BUGS","text":"

    Report bugs to https://github.com/AllStarLink/ASL3/issues

    "},{"location":"mans/asl-setup-dkms-mok/#copyright","title":"COPYRIGHT","text":"

    Copyright (C) 2024 Jason McCormick and AllStarLink under the terms of the AGPL v3.

    "},{"location":"pi/cockpit-basics/","title":"Cockpit Basics","text":"

    The Cockpit web console is a web-based graphical interface By using it, you can perform system administration tasks, such as inspecting and controlling services, managing storage, configuring networks, analyzing network issues, and inspecting logs. Of course, it's also very handy for administering Asterisk on your AllStarLink Pi Appliance!

    "},{"location":"pi/cockpit-basics/#what-is-the-cockpit-console","title":"What is the Cockpit Console","text":"

    What is the Cockpit Console

    The Cockpit Console is a web-based interface designed for managing and monitoring your AllStarLink Pi Appliance.

    The Cockpit Console enables you to perform a wide range of administration tasks, including:

    Managing AllStarLink, Asterisk, and its related applications\nManaging services\nManaging user accounts\nManaging and monitoring system services\nConfiguring network interfaces\nReviewing system logs\nUpdating software\n

    The Cockpit Console uses the same system concepts as you would use in a terminal, and actions performed in a terminal are immediately reflected in the Cockpit Console. That is to say, you can use Cockpit and an SSH connection interchangeably at your desire.

    You can monitor the logs of application. In addition, you can change the settings directly in the web console or through the terminal.

    Throughout this documentation a host named node63001.local is used as the example host. Substitute that name with your own node's name as it was set during the imaging process.

    "},{"location":"pi/cockpit-basics/#logging-in-to-the-web-console","title":"Logging In to the Web Console","text":"

    1. In your web browser, enter the following address to access the web console: https://node63001.local:9090

    2. The browser may display a warning. Accept the security exception to proceed with the login.

    3. Enter the username and password set during the imaging process.

    4. Click Log In.

    "},{"location":"pi/cockpit-basics/#administrative-access","title":"Administrative Access","text":"

    The Web Console has a concept of Administrative access. While it does remember the access level across logins, to access certain functions you may need to enable Administrative access from Limited access. Do do that:

    1. Click on the bar marked Limited Access to enable admin-level access.

      You may be prompted to tnter the password again. Do so and then click Authenticate

    2. Click Close on the box reporting administrative access has been granted.

    "},{"location":"pi/cockpit-basics/#shutting-down-the-appliance","title":"Shutting Down the Appliance","text":"

    You can use the web console to shut down the system.

    1. Log into the Web Console.

    2. (optional) Click on the bar marked Limited Access to enable admin-level access.

    3. (optional) Click Close on the box reporting administrative access has been granted.

    4. Click on the downward arrow next to Reboot and then select Shutdown

    5. Change the Delay dropdown to No delay and click Shut down

    6. A Disconnected message will appear. The node will shutdown is about 15-30 seconds, after which power may be removed.

    "},{"location":"pi/cockpit-basics/#rebooting-the-appliance","title":"Rebooting the Appliance","text":"

    1. Log into the Web Console.

    2. (optional) Click on the bar marked Limited Access to enable admin-level access.

    3. (optional) Click Close on the box reporting administrative access has been granted.

    4. Click on Reboot

    5. Change the Delay dropdown to No delay and click Reboot

    6. A Disconnected message will appear. The node will reboot and be available again in about 15-30 seconds. After that, one can click Reconnect.

    "},{"location":"pi/cockpit-basics/#configuring-time-on-the-appliance","title":"Configuring Time on the Appliance","text":"

    In general, the appliance should be left in the default setting of using the NTP service to automatically set the time. However you may want to change the Timezone. To change the time:

    1. Log into the Web Console.

    2. (optional) Click on the bar marked Limited Access to enable admin-level access.

    3. (optional) Click Close on the box reporting administrative access has been granted.

    4. Click on Overview and scroll down to Configuration

    5. Click on highlighted time next to System time

    6. Change the Time zone drop-down to the desired zone.

    7. Click Change.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-console/","title":"Terminal / Console","text":"

    The Cockpit Web Console provides a feature-complete terminal interface in the web browser without requiring setup and configuration of an SSH client. This allows administration of the node from anywhere!

    Use of the web console Terminal and an SSH client is fully interactive and interchangeable.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Terminal in the web console menu on the left.

    3. You will be connected to a terminal of the user that you logged in with.

    4. To execute commands as root, either prefix the command with sudo:

    5. One can enter a root session completely by issueing the command sudo -s:

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-firewall/","title":"Managing the Firewall","text":"

    A firewall is a way to protect nodes from any unwanted traffic. It enables users to control incoming network traffic on host machines by defining a set of firewall rules. These rules are used to sort the incoming traffic and either block it or allow it through. In the AllStarLink Pi APpliance, the firewalld service with the nftables back end functions as the firewall. Through the web console, you can configure firewalld.

    "},{"location":"pi/cockpit-firewall/#adding-an-inbound-port","title":"Adding an Inbound Port","text":"

    The firewall is enabled by default with a reasonable set of default rules. The most common reasons to edit the port list would be to change the IAX2 port to be outside the range of 4560-4580 (default is 4569) or to change the voter port (default 1667 in ASL3, might need 667 for legacy reasons).

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. Click Edit rules and zones in the Firewall panel.

    4. Click Custom ports

    5. Enter the TCP or UDP port or ports to be permitted in the TCP or UDP fields respectively. Multiple ports can be separated with commands and a range of ports can be specified with a hyphen. It is highly suggested to include a Description for the port. For example, if adding an alternative IAX2 port on 14569:

    6. Click Add ports

    7. The change is immediate.

    "},{"location":"pi/cockpit-firewall/#removing-an-inbound-port","title":"Removing an Inbound Port","text":"

    The firewall is enabled by default with a reasonable set of default rules. The most common reasons to edit the port list would be to change the IAX2 port to be outside the range of 4560-4580 (default is 4569) or to change the voter port (default 1667 in ASL3, might need 667 for legacy reasons).

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. Click Edit rules and zones in the Firewall panel.

    4. Next to the service to delete, click on the three vertical dots and select Delete. The change is immediate.

    "},{"location":"pi/cockpit-firewall/#enabling-and-disabling","title":"Enabling and Disabling","text":"

    The firewall is enabled by default with a reasonable set of default rules. Unless you are familiar with networking, DO NOT disable the firewall or make changes to it.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. To enable the firewall, slide the toggle to the right. This is the default setting and should not be changes unless you're 100% sure of what you are doing.

    4. To disable the firewall, slide the toggle to the left. Donot disable the firewall unless you're 100% sure of what you are doing.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-get-started/","title":"Pi Appliance","text":"

    AllStarLink 3 features a modern appliance of AllStarLink while retaining the ubiquitous Raspberry Pi experience that's well-documented and familiar to the amateur radio community worldwide.

    "},{"location":"pi/cockpit-get-started/#cockpit-web-admin-interface","title":"Cockpit Web Admin Interface","text":"

    The AllStarLink Pi Appliance includes the user-friendly Cockpit system for ease of administration. Cockpit is access on port 9090 of the appliance. In these examples the hostname node63001 is used and should be replaced with the name you set during setup. If the name node63001 was used at setup, then pointing a browser to https://node63001.local:9090/ will bring up the Cockpit interface. Note: The hostname only works when your node and your PC are on the same LAN. Otherwise use the IP or dns name you assign. Login is the username and password that was setup during the Pi imaging process.

    The first connection will report a message that \"Your connection isn't private\". For the Pi appliance, this is acceptable. Click on Advanced and then Continue to node63001.local (unsafe). This only must be done the first time.

    Enter the username and password configured during the imaging process above. Click Log in

    Cockpit is fairly intuitive to use, but here's are a few pointers to help get started:

    "},{"location":"pi/cockpit-network/","title":"Network Configuration","text":"

    The Networking system tool in the web console permits viewing the status of the network as well as making network configuration changes to the appliance.

    "},{"location":"pi/cockpit-network/#observing-network-status-and-configuration","title":"Observing Network Status and Configuration","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. At the top of the screen will be the network data statistics for transmitted and receiving network traffic. This information can be helpful if there are questions about network bandwidth utilization:

    4. The next panel down shows the current configuration of the present network adapters:

    5. Finally, at the bottom is the logs about the current state of the network:

    "},{"location":"pi/cockpit-network/#changing-adapter-setting","title":"Changing Adapter Setting","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Networking in the web console menu on the left.

    3. Scroll down to the Interfaces section

    4. Click on the Name of the interface

    5. Make desired changes

    6. Changes take place immediately. Be cautious when changing IP addressing because you may accidentally remove your ability to access to the console.

    "},{"location":"pi/cockpit-network/#changing-wifi-settings","title":"Changing WiFi Settings","text":"

    Unfortunately, Cockpit does not have an interface for managing WiFi networks with in it. Currently WiFi network changes must be made from the Web Console Terminal screen.

    "},{"location":"pi/cockpit-network/#listing-existing-wifi-networks","title":"Listing Existing WiFi Networks","text":"

    List the existing WiFi network(s) with nmcli device wifi list:

    "},{"location":"pi/cockpit-network/#add-a-wifi-network","title":"Add a WiFi Network","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Terminal in the web console menu on the left.

    3. At the prompt enter the command:

      nmcli device wifi connect MYNETWORK password MYPASSWORD\n

      where MYNETWORK is the name (\"SSID\") and MYPASSWORD is the password (\"Key\"). To join the network \"RadioSite\" with the password \"RF1234\":

      nmcli device wifi connect RadioSite password RF1234\n

      If there is a space in the name or password, you must use quotes around the element with the space in it.

    4. If the network is not currently active (i.e. adding a network where the appliance will be in the future) or the SSID is hidden, adding is slightly more complicated. Again, substitute MYNETWORK and MYPASSWORD with as appropriate.

      nmcli conn add type wifi con-name MYNETWORK ifname wlan0 ssid MYPASSWORD\nnmcli conn modify MYNETWORK wifi-sec.key-mgmt wpa-psk wifi-sec.psk MYPASSWORD\n

    Note: DO NOT use the raspi-config tool to attempt to set the Wifi network and password. This will break the appliance's networking.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-services/","title":"Services Management","text":"

    The Cockpit Web Console allows the management of all of the running services. You can activate or deactivate services, restart or reload them, or manage their automatic startup.

    "},{"location":"pi/cockpit-services/#start-stop-and-restart-services","title":"Start, Stop, and Restart Services","text":"

    This procedure stops or starts system services using the web console interface. You can filter the services by name or description and also by Enabled, Disabled, or Static automatic startup. The interface shows the current state of the service and its recent logs.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Services in the web console menu on the left. The default tab for Services is System Services. If you want to manage targets, sockets, timers, or paths, switch to the respective tab in the menu on top. To open service settings, click on a selected service from the list. You can tell which services are active or inactive by checking the State column.

    3. To start an inactive service, click the Start button.

    4. To deactivate an active service, click the Stop button

    5. To restart a service, click the Restart button

    "},{"location":"pi/cockpit-services/#seeing-status-and-recent-logs-of-a-service","title":"Seeing Status and Recent Logs of a Service","text":"

    This procedure stops or starts system services using the web console interface. You can filter the services by name or description and also by Enabled, Disabled, or Static automatic startup. The interface shows the current state of the service and its recent logs.

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Services in the web console menu on the left. The default tab for Services is System Services. If you want to manage targets, sockets, timers, or paths, switch to the respective tab in the menu on top. To open service settings, click on a selected service from the list. You can tell which services are active or inactive by checking the State column.

    3. Scroll down to see Service logs. The botton View All Logs can be clicked to review an extensive output of the service's logs.

    Note: Some content copied from Managing systems using the RHEL9 web console which is released under the Creative Commons Attribution\u2013Share Alike 3.0 Unported license (\"CC-BY-SA\")

    "},{"location":"pi/cockpit-updates/","title":"Software Updates","text":"

    The Software updates tool in the web console is how to install package updates to the AlLStarLink Pi appliance. This includes updating AllStarLink software and its associated items.

    "},{"location":"pi/cockpit-updates/#checking-for-and-installing-updates","title":"Checking For and Installing Updates","text":"

    1. Log in to the web console with administrator privileges. For details, see Cockpit Basics.

    2. Click Software Updates in the web console menu on the left.

    3. Click on the cycle icon in the upper right of the panel.

    4. Available updates will be displayed. Click on Install all updates

    5. The updates will start. Depending on the quantity of updates, the speed of the Pi hardware, and how long certain installs take, this may take quite awhile to copmlete. For example, if there is a Linux kernel update, then the Dahdi package will be recompiled for the new kernel which can take up to 10 minutes. Be patient while updates run. You can observe the updates by clicking on View update log.

    6. When the updates are complete, you will be prompted to reboot the system. In general, it is advised to reboot the system after updates, however that is not necessarily required for most updates. Kernel updates and Dahdi updates DO require a reboot.

    7. After the updates are completed, returning to the Software Updates screen should show that the software is up to date.

    "},{"location":"user-guide/","title":"ASL3 User Guide","text":"

    This is the AllStarLink version 3 User's Guide. No doubt you're you've noticed this is not a Wiki. The ASL Wiki still exists, largely to document ASL1 and ASL3, while this manual covers ASL3 specifics and is the future home for all ASL documentation.

    Did you see the \"Next\" and \"Previous\" menu items? Those will page to the next section of this document in the order of the bullets below. We think you'll find this an orderly way to learn about ASL3. Lets get started.

    "},{"location":"user-guide/asl-cmds-tools/","title":"ASL Commands and Tools","text":"

    The following commands and tools are the primary AllStarLink ecosystem. Each entry contains a description of the command, link to the documentation, and what package provides the command or tool.

    Command Description Documentation Package asterisk The main console for Asterisk and app_rpt Normally invoked with asterisk -r asl3-asterisk asl-menu The main text-based menu for configuring ASL Menu Use asl3 asl3-update-nodelist Primary engine to update the /var/lib/asterisk/rpt_extnodes database like for the node dictionary. Node Resolution asl3-update-nodelist asl3-update-astdb Universal engine to provide astdb.txt to various applications that require the old node name database. Use this rather than a special version for every third-party app (e.g. Supermon, Node Remote, etc.) Node Resolution asl3-update-nodelist allmon3-passwd Manage the user database and passwords for Allmon3 Allmon3 User Management allmon3 asl-find-sound Utility to help locate compatible sound devices manpage asl3 asl-node-lookup Utility to test node lookup when troubleshooting Asterisk/app_rpt connectivity issues manpage asl3 asl-play-arn Tool to play Amateur Radio Newsline manpage asl3 asl-repo-switch Utility to change package release streams for AllStarLink Beta Testing asl3 asl-say Tool for basic statements into Asterisk like the sime manpage asl3 asl-show-version Utility to report the package version of all related ASL3 software. Normally used in bug reports and support requests. N/A asl3 asl-tts Text-to-speech engine for AllStarLink Text to Speech asl3-tts sa818 CLI tool for configuration SA818/DRA818-based radio modules. Use `sa818-menu in most cases. SA818 Page asl3 sa818-menu Menu for configuring SA818/DRA818-based radio modules SA818 Page asl3"},{"location":"user-guide/betatest/","title":"Beta Testing","text":"

    Beta testing is the final phase of software testing before releasing a product to the general public. During this stage, a group of users representing the target audience validates the product. They explore its features, uncover bugs, and provide feedback.

    AllStarLink releases software packages to a beta channel on a periodic basis to address bug fixes and feature enhancements. The beta channel of packages is available to all users. However, please only select to run packages from the beta channel if you meet the following criteria:

    1. If something goes wrong, your can easily recover your node from backups or other saved configuration.

    2. You have generalized Linux skills such as understanding basic commands, ability to use the shell terminal of Linux, ability to copy/paste output data into bug reports, etc.

    3. You acknowledge that a beta release may have other bugs, incomplete fixes, or may in rare cases not work right at all for your configuration and that you can live with this.

    4. You may need to edit configuration files to fix issues or test things or to roll back to previous versions of software.

    If you cannot adhere to the above four conditions, you should not use the beta channel software.

    "},{"location":"user-guide/betatest/#enabling-beta-channel-software","title":"Enabling Beta Channel Software","text":"

    Enabling beta channel software is done with the asl-repo-switch command as root. Set the channel to \"beta\" using asl-repo-switch -l beta. For example:

    # asl-repo-switch -l beta\nSetting level beta... DONE\nRun 'apt update' to refresh the package repo cache.\n
    "},{"location":"user-guide/betatest/#disabling-beta-channel-software","title":"Disabling Beta Channel Software","text":"

    Disabling beta channel software is done with the asl-repo-switch command as root. Set the channel to \"main\" using asl-repo-switch -l main. For example:

    # asl-repo-switch -l main\nSetting level main... DONE\nRun 'apt update' to refresh the package repo cache.\n
    "},{"location":"user-guide/betatest/#installing-updates-using-beta-channel","title":"Installing Updates Using Beta Channel","text":"

    The most common case to use the beta channel for software packages is to test a proposed fix to a problem you are having. In those cases, the following commands should be run one at a time from a terminal/console as root (using sudo or sudo -s):

    sudo asl-repo-switch -l beta\nsudo apt update\nsudo apt upgrade -y\nsudo asl-repo-switch -l main\n

    This will do a one-time installation of the packages in the beta channel and then switch back to main.

    "},{"location":"user-guide/betatest/#resetting-installation-of-asl-back-to-production","title":"Resetting Installation of ASL Back to Production","text":"

    To reset your packages for ASL3 back to the production-release main versions:

    For Pi Appliances: 

    sudo asl-repo-switch -r\nsudo apt update\nsudo apt install -y --reinstall asl3 asl3-asterisk asl3-asterisk-config \\\n  asl3-asterisk-modules asl3-menu asl3-update-nodelist \\\n  asl3-pi-appliance allmon3\n

    For Debian 12 general installations: 

    sudo asl-repo-switch -r\nsudo apt update\nsudo apt install -y --reinstall asl3 asl3-asterisk asl3-asterisk-config \\\n  asl3-asterisk-modules asl3-menu asl3-update-nodelist\n

    For DAHDI Kernel packages after doing the above, as necessary: 

    sudo apt install -y --reinstall dahdi-linux dahdi-dkms\n

    Note that you may need to merge or fix configurations when rolling back from beta software to production. This is unusual but possible.

    "},{"location":"user-guide/install/","title":"Debian 12 Install","text":"

    These instructions are for installing ASL3 on general purpose operating systems manually. For installation on Raspberry Pis, consider using the AllStarLink 3 Pi Appliance

    Architecture Support

    Currently, the project does not support armv7l/armhf platforms because all known uses of AllStarLink is on hardware which supports the Bookworm arm64 distribution such as Raspberry Pi 3, 4, 5, and Zero 2 W. If you have a platform that can use armv7l/armhf 32-bit packages only please file an issue at ASL3 on GitHub.

    Asterisk/app_rpt on VMs/Cloud

    Asterisk/app_rpt works fine on virtual machines in a cloud environment (often terms \"VPS\"). However Asterisk/app_rpt requires certain kernel modules to function properly and these are provided by ASL3 in the dkms-dahdi package. There are, however, two situations where Asterisk/app_rpt will not work:

    1. When the VPS provider is using a customized Debian using the \"cloud\" kernel. If the output of uname -r refers to the kernel as something similar to \"6.1.0-12-cloud-x86_64\" then you must follow this procedure prior to installing ASLv3.

    2. When the virtual machine is using an emulated Linux kernel. Microsoft Windows Subsystem for Linux (WSL2) is a good example of this situation. The kernel is custom and not related to the operating system distribution. Asterisk/app_rpt cannot work in this type of setup.

    "},{"location":"user-guide/install/#system-requirements","title":"System Requirements","text":"

    The following are the system requirements for an ASL3 system:

    Minimum Required Recommended CPU/Platform 1 CPU, 64-bit, x86_64 (amd64) or ArmV8 (arm64) 2 - 4 CPUs depending on the number of hardware devices connected to the system Memory 512M 2 G Storage 8G (for OS + software) -

    UEFI / SecureBoot

    For x86_64/amd64 platforms, it is recommended to disable UEFI and SecureBoot if you do not need those features and your platform supports a \"legacy BIOS\" mode. While they are a good security feature, given that AllStarLink v3 requires building a kernel module, it adds likely-undesired complexity for most ASL users. If you need or want to use UEFI/SecureBoot see the advanced topic document.

    "},{"location":"user-guide/install/#os-install","title":"OS Install","text":"

    You\u2019re going to start off by installing a new Debian 12 OS on your PC computer or virtual machine. There are instructions all over the internet that detail how to install Debian 12. Briefly you:

    Note: When setting up users and passwords you may be prompted to set the password for the 'root' account. If you do not provide a password then the root account will be disabled and the system's initial user account (that will requested on the next screen) will be given the power to become root using the \"sudo\" command. If you do provide a password then the \"sudo\" command will not be installed on your system. Our recommendation is to NOT set the 'root' password and rely on using the \"sudo\" command only when needed.

    "},{"location":"user-guide/install/#allstarlink-package-repo-install","title":"AllStarLink Package Repo Install","text":"

    Once your Debian system is up and running, install the ASL package repositories:

    cd /tmp\nwget https://repo.allstarlink.org/public/asl-apt-repos.deb12_all.deb\nsudo dpkg -i asl-apt-repos.deb12_all.deb\nsudo apt update\n
    "},{"location":"user-guide/install/#asl3-packages-install","title":"ASL3 Packages Install","text":"

    Now the packages may be installed and updated directly from the AllStarLink package repository:

    sudo apt install asl3\n

    This will install the complete ASL3 system including all of the Asterisk app_rpt-enabled packages (asl3-asterisk, asl3-asterisk-config, asl3-asterisk-doc, asl3-asterisk-modules), the Dahdi kernel module (dahdi-dkms, dahdi-linux, dahdi-source), the needed development tools to keep the kernel module updated, and the asl3-menu.

    It's important to note that the new packaging format will allow for easy, automated updates of kernels through the standard apt upgrade process. There is no longer any reason to hold back kernel upgrades with the ASL3 packaging.

    The following packages are also available and may be installed separately:

    Allmon3 - The updated web interface to AllStarLink which includes strong support for mobile devices and screens of all sizes and shapes. See the Allmon3 page for details.

    ASL3 Nodelist Updater - An updated node service to maintain a local copy of the AllStarLink node database (rpt_extnodes). While, in general, the preferred method of node lookup is DNS, some installations with slow Internet or DNS servers that have very long cache timers may benefit from this method of node lookup instead. Install with sudo apt install asl3-update-nodelist.

    "},{"location":"user-guide/install/#node-configuration","title":"Node Configuration","text":"

    Next step is to configure the node settings. It is recommended to use the asl-menu command to manage the configuration for common use cases. See ASL3 Menu for details. YouTuber Freddie Mac has a nice ASL3 RPi installation and configuration video. See the part where the asl-menu is shown.

    "},{"location":"user-guide/install/#docker","title":"Docker","text":"

    Docker support will be coming in the near future.

    "},{"location":"user-guide/known-issues-diffs/","title":"Known Issues & Differences With Legacy Versions","text":"

    As discussed in What's New?, AllStarLink v3 is a gigantic leap forward in platform and software ecosystem. Many things are different in modern versions of Linux, Asterisk, and more. This page documents permanent differences with ASL3 relative to legacy versions as well as any known issues that must be fixed in ASL3.

    "},{"location":"user-guide/known-issues-diffs/#differences-with-legacy-versions","title":"Differences with Legacy Versions","text":""},{"location":"user-guide/known-issues-diffs/#serial-ports-available-by-default","title":"Serial Port(s) Available by Default","text":"

    On the ASL3 Pi Appliance, the system comes pre-configured for /dev/serial0 (formerly /dev/ttyAMA0) accessibility. That means that Bluetooth and the default serial console are disabled. Any directions requiring editing of config.txt or cmdline.txt are unnecessary with the ASL3 appliance.

    "},{"location":"user-guide/known-issues-diffs/#pi-dev-entry-changes","title":"Pi /dev Entry Changes","text":"

    As ASL3 is based on Debian 12, users with Raspberry Pi devices must note that the serial port on the Pi header is now /dev/serial0 rather than the historical /dev/ttyAMA0. If you are following directions for Pi serial port operations, such as programming an SA818/DRA818-based radio hat or a SHARI node, use /dev/serial0 in place of the /dev/ttyAMA0 reference.

    "},{"location":"user-guide/known-issues-diffs/#voterrtcm-default-port","title":"Voter/RTCM Default Port","text":"

    Modern installations of Asterisk runs as the unprivileged asterisk user rather than as root. Standard Linux convention prohibits non-root users to listen on a TCP port below 1024. The default port for Voters/RTCMs has been changed to 1667 when previously it was 667. If voter port changes are difficult for the environment, see Incompatible Changes in ASL3 for other potential workarounds.

    "},{"location":"user-guide/known-issues-diffs/#known-issues","title":"Known Issues","text":"

    The following issues are currently known to exist in AllStarLink 3 and, where possible what the workarounds are.

    "},{"location":"user-guide/known-issues-diffs/#dtmf-linking-dns-lookup-timeouts","title":"DTMF Linking + DNS Lookup Timeouts","text":"

    There are intermittent cases of linking problems in the following situation:

    1. When a linking command (e.g. *3) is sent in DTMF; AND
    2. Certain network configurations or situations do not resolve DNS \"fast enough\"

    This issue is fully described in app_rpt GitHub issue #392. In summary, if the DTMF entry, decoding, and DNS resolution for a link command collectively takes longer than 3 seconds, linking can fail in silent and indeterminate ways.

    If this problem is occurring consistently, changing the node lookup method to \"file\" works around the issue until updated code to resolve this problem can be developed and tested:

    1. Install the asl3-update-nodelist package (not needed for those running the ASL3 Pi Appliance):

      sudo apt install -y asl3-update-nodelist\n

    2. Edit /etc/asterisk/rpt.conf. Somewhere at or around line 10 (depending on file customizations) you will need to change the node_lookup_method to file instead of both. After your change the file shoud look like:

      [general]\nnode_lookup_method = file       ;method used to lookup nodes\n                                ;both = dns lookup first, followed by external file (default)\n                                ;dns = dns lookup only\n                                ;file = external file lookup only\n

      This file can be updated using asl-menu \"Expert Configuration Menu\" / \"Edit rpt.conf file\" or your favorite editor.

    3. Restart Asterisk using asl-menu \"Node Settings\" / \"Restart Asterisk\" or systemctl restart asterisk.

    Once the fix is completed and made available, it will be announced on the ASL Community and you should change node_lookup_method to both. Eventually the \"file\" method will be retired.

    "},{"location":"user-guide/known-issues-diffs/#resize2fs_once-error","title":"resize2fs_once \"Error\"","text":"

    There are intermittent cases of errors on the screen or in the system logs about a failure of a service named resize2fs_once.service after the final first boot upon installation. The error may report that it \"Failed to start\" or \"timed out\". If the / partition has been properly resized - which has been the case in every known occurrence of the error - then there is no action to take and the issue will not appear on subsequent reboots.

    A properly resized / should be a bit smaller than the full size of the SD card or USB drive used with the device. In Cockpit, look at the Storage tab:

    In this example, / is a 31G partition on a 32G SD card.

    "},{"location":"user-guide/known-issues-diffs/#shutdownrestart-issues-with-ezstreambroadcastify-enabled","title":"Shutdown/Restart Issues with EZStream/Broadcastify Enabled","text":"

    With the configuration for Broadcastify enabled, shutting down or restarting Asterisk is not quick or always reliable. When issuing systemctl stop or systemctl restart for the asterisk.service unit it will take upwards of 90 seconds to properly terminate and exit/respawn. Make sure any lame and ezstream processed are killed and not orphaned.

    # ps auxww | grep asterisk\nasterisk 1443572  0.0  0.0   2416  1024 ?        S    10:13   0:00 /bin/sh -c /usr/bin/lame --preset cbr 16 -r -m m -s 8 --bitwidth 16 - - | /usr/bin/ezstream -qvc /etc/ezstream.xml\nasterisk 1443573  0.1  0.1  11920  5120 ?        S    10:13   0:00 /usr/bin/lame --preset cbr 16 -r -m m -s 8 --bitwidth 16 - -\nasterisk 1443574  0.0  0.2  55232 10240 ?        S    10:13   0:00 /usr/bin/ezstream -qvc /etc/ezstream.xml\n

    If Asterisk is shutdown, the processes above must be stopped with a kill -9 PID command.

    "},{"location":"user-guide/menu/","title":"ASL3 Menu","text":"

    Once you have installed ASL3 on your system you will want to configure your node. The ASL3 menu will help you get started with many common configurations including nodes with USB audio interfaces and \"hub\" nodes. The ASL3 menu will also help you maintain your system.

    Your system may have been setup to automatically start the ASL3 menu as soon as you have logged in. If not, you can access the ASL3 menu by executing the following command from the Linux shell :

    asl-menu\n

    Note: if you are not logged in as the \"root\" user you will need to use

    sudo asl-menu\n

    When the ASL3 menu is running you will be presented with options to configure a node, access the Asterisk CLI, backup (or restore) your ASL settings, access the Linux CLI, perform some basic diagnostics, and perform some common system actions.

    "},{"location":"user-guide/menu/#node-settings","title":"Node settings","text":"

    For new installs you will want to select the ASL3 \"Node Settings\" menu option. This is where you will configure the basic settings needed to get your new node on the air. You can return to this menu later to review and/or update your node settings.

    The \"Node Settings\" menu allows you update key node settings including :

    The \"Node Settings\" menu also allows you to add (or remove) additional nodes.

    Note: you can also use the node-setup command to access this menu.

    "},{"location":"user-guide/menu/#asterisk-cli","title":"Asterisk CLI","text":"

    The Asterisk CLI provides direct access to the heart of what makes an AllStar node. You might use the CLI for testing, troubleshooting, or for controlling your node. You should know that many of the commands require a more in-depth knowledge of Asterisk. The good news is that many of the commonly used commands can be executed from the \"Diagnostics Menu\" or with one of the web management/monitoring applications (e.g. Allmon3).

    "},{"location":"user-guide/menu/#backup-and-restore","title":"Backup and Restore","text":"

    The ASL3 \"Backup and Restore\" menu option provides a simple option to backup (and restore) your AllStar and Asterisk configuration. The backup archives are stored locally on your system and optionally in the cloud.

    We encourage you to \"backup\" your configuration, especially before and after you make changes.

    Note: you can also use the asl-backup-menu command to access this menu.

    "},{"location":"user-guide/menu/#diagnostics","title":"Diagnostics","text":"

    The ASL3 \"Diagnostics\" menu option allows you to perform some simple/rudimentary checks of your system and node. These include basic internet connectivity checks and reporting of your AllStar registrations. We also provide a way to stop, start, and restart some of the AllStar and Asterisk services that run on your system.

    Note: you should know that these diagnostics are not exhaustive. With a good connection to the internet and a valid configuration all \"should be\" OK. If the provided diagnostics don't flag an issue then you will need to dig deeper. Be aware of changes you make to your system and configuration. Be aware of software packages that you install. Be aware of \"upgrades\". If you need help, reach out to the AllStarLink Community.

    "},{"location":"user-guide/menu/#expert-configuration","title":"Expert Configuration","text":"

    The ASL3 \"Expert Configuration Menu\" option will allow you to make changes to some of the AllStar and Asterisk configuration files (in the \"/etc/asterisk\" directory). But, don't worry about that advanced stuff just yet.

    "},{"location":"user-guide/menu/#bash-shell","title":"Bash shell","text":"

    The \"bash\" shell provides access to the Linux command line interface, also known as the Linux CLI. This is for experts or if you have been provided with CLI instructions.

    "},{"location":"user-guide/passwords/","title":"Logins and Passwords","text":"

    Yes, there are many \"different\" logins and passwords associated with AllStarLink, the ASL3 software, and the related applications. A typical ASL3 server / node will directly or indirectly use the following login/password pairs :

    "},{"location":"user-guide/passwords/#allstarlink-website-wwwallstarlinkorg","title":"AllStarLink website (www.allstarlink.org)","text":"

    Th AllStarLink website is where you register for an AllStarLink account, add ASL servers, and request ASL nodes.

    "},{"location":"user-guide/passwords/#linux","title":"Linux","text":"

    Your desktop, laptop, Raspberry Pi, or Virtual Machine will be running the Linux operating system (e.g. Debian). To access the operating system on your OS you will need a login \"user\" and \"password\".

    You will use the Linux \"user\" and \"password\" when logging in to your system :

    You may have heard about the Linux \"root\" account. The root account is also called the super-user; it is a login that bypasses all security protection on your system. The root account should only be used to perform system administration, and only used for as short a time as possible. With ASL3, we are trying to limit our usage of the \"root\" account.

    Note: there is NO DEFAULT USER included in the ASL3 Raspberry Pi Appliance image.

    "},{"location":"user-guide/passwords/#asterisk-manager-interface-ami","title":"Asterisk Manager Interface (AMI)","text":"

    Your AllStarLink server system runs as process named \"asterisk\" that is the heart of your node(s). The Asterisk Manager Interface, AMI, is the standard management interface into your Asterisk server.

    The AMI configuration, stored in \"/etc/asterisk/manager.conf\", includes \"user\" categories and associated \"secret\" passwords. For example :

    [admin]\nsecret = my-AMI-password\n...\n

    This \"user\" (admin) and \"secret\" (my-AMI-password) would need to be specified in the configuration file for any applications that use AMI to interact with Asterisk.

    "},{"location":"user-guide/passwords/#web-applications-eg-allmon3","title":"Web Applications (e.g. Allmon3)","text":"

    Many will install one (or more) web applications that monitor/control their ASL nodes. These applications can provide access to many users on the web and those users may have no relation to those who setup/administer/use your systems. As such, these applications typically maintain their own list of users and associated passwords.

    Please refer to the web application manual for information on how to manage who is allowed access to the web pages.

    Also, as mentioned above, many of these applications use AMI to interact with your Asterisk and your nodes. The web application manual should have information on where to specify the AMI credentials.

    "},{"location":"user-guide/pi-detailed/","title":"Raspberry Pi Install","text":"

    The Raspberry Pi image includes the following to make a complete AllStarLink system:

    You must use the Raspberry Pi Imager even if you prefer another imager and carefully follow the instructions below. Failure to do so will result in a system that won't boot or that you will not be able to login to.

    You will install an image on a microSD card and go. This is the simplest install. For most nodes, the menus will walk you through the setup. Note: a minimum SD card size of 4G is required and it is recommended to use at least an 8G SD card.

    "},{"location":"user-guide/pi-detailed/#pi-appliance-requirements","title":"Pi Appliance Requirements","text":"

    The following are the system requirements for the Pi Appliance

    Required Recommended Hardware Arm v8 64-bit CPU Raspberry Pi 3, 4, 5, Zero 2 W, 400 Raspberry Pi 4B or 5B Memory Minimum 512M 4G Storage Minimum 4G or larger 8GB Class 10

    The ASL3 Appliance been tested on Raspberry Pi 3, 4, 5, and Zero 2W.

    "},{"location":"user-guide/pi-detailed/#step-by-step-pi-appliance-setup","title":"Step-by-Step Pi Appliance Setup","text":"

    These directions are specific for Windows by example , but in general, should work the same for the same tool set on MacOS and Linux. For each screenshot below, click on the image for a larger version.

    Note: For this example and all documentation, the hostname used shall be node63001 and the node number is 63001. Replace \"63001\" with your node number as you proceed through these directions.

    1. If you do not already have it installed, install the Raspberry Pi Imager.

    2. Download the latest release image. This will be named allstar3-arm64-X.Y.Z.img.xz where \"X.Y.Z\" is the version. The version will be something like 3.0.10. Save it to your local Downloads directory.

    3. Launch Raspberry Pi Imager from the start menu.

    4. Click on CHOOSE DEVICE and choose the type of Pi hardware on which AllStarLink 3 Appliance is being installed.

    5. Click on CHOOSE OS. Scroll to the bottom of the list that appears and select Use custom.

    6. A Select image dialog box will appear. Navigate to the Downloads folder and select the image downloaded in step 2. It will be named something like allstar3-arm64-3.0.1.img.xz. Then click Open.

    7. Connect the SD card or the SD card in a USB adapter to the computer. Then click on CHOOSE STORAGE. An option will be presented named something such as \"Mass Storage Device USB Device - 16.0 GB\". Click on that entry.

    8. Click NEXT

    9. A box labelled \"Use OS customisation?\" will appear. Click on EDIT SETTINGS

    10. Check the box next to Set hostname and enter a hostname for the device to appear as on the network. In this example, node63001 is used. Unless you have another naming convention for AllStarLink nodes, use the nodeNNNNN format.

    11. Check the box next to Set username and password and enter a username and password. It is recommended to set the username to 'asl' if you have no strong feeling about the username. Choose a good password and record the password somewhere safe. NOTE: There is NO DEFAULT USER for the image. Failure to set an account will require a reimage of the SD card with the proper settings.

    12. If the device will be connected to WiFi, check the box \"Configure wireless LAN\" and then enter the name of the wireless network in SSID and the password for that network in Password. Change the \"Wireless LAN country\" to US or another country as appropriate. Setting this is not necessary if the Pi will use a network cable rather than wireless.

    13. Check the box next to Set locale settings and change the \"Time zone\" to the desired region. The timezone settings will control any time-based announcements and schedule jobs. After completing steps 10-13, the customization should look something like:

    14. Click SAVE in \"OS Customisation\".

    15. Click YES for \"Use OS customisation?\"

    16. Click YES to continue to write the image to the SD card.

    17. Depending on the speed of the computer and the type of SD card one will have time for a beverage of their choice. When the write is complete, eject the card, remove the card from computer and insert it into the Pi. If using a USB adapter for the SD card, remove the SD card from the adapter and insert the card into the Pi. The Pi will not use the SD card in the USB adapter.

    18. Power on the Pi. Wait approximately 2 to 5 minutes for the Pi to boot and perform the various first boot tasks. Depending on the type of a Pi board used, this process could be lengthy. If there is a screen connected to the Pi, it will be noted that Pi will reboot several times. This is normal and expected.

    19. (Optional) Network connectivity may be tested using the command to ping the hostname set in step 10. For example, if the hostname set was \"node63001\" then do ping -4 node63001.local from a command prompt or PowerShell window. When you get a response, the host is connected to the network.

    20. Open the browser on your computer and navigate to https://nodeNNNNN.local. In this example the name was node63001 then one would enter https://node63001.local. Your Pi and your computer must be on the same LAN for this to work. Use the IP address rather than the hostname if the Pi is on a remote network or the hostname is unreachable. Depending on your network and browser, one may have to enter the name several times until the page displayed. The first connection will report a message that \"Your connection isn't private\". For the Pi appliance, this is acceptable. Click on Advanced and then Continue to node63001.local (unsafe). This only must be done the first time.

    21. The AllStarLink Launchpad will appear. Clicking on Web Admin Portal will take one to the web-based administration portal that includes a web-based SSH client for using asl-menu and other commands. Clicking on Node Links will take one to the Allmon3 interface. Clicking on ASL Manual will take one to the top of this manual.

    22. To setup the new node, click on Web Admin Portal. Similar to step 20, click Advanced and Continue to node63001.local (unsafe) to see the logon box.This only must be done the first time. Enter the username and password configured during the imaging process above. Click Log in

    23. The Overview tab is displayed. Using the Web Admin Portal (Cockpit) with AllStarLink is covered in the manual. At the top, click on *Turn on administrative access

      Enter the password again when prompted and click Authenticate

    24. The web-based SSH console appears as Terminal at the bottom of the left scrollable menu. Click it and SSH console will appear:

    The Terminal window is a complete interface to AllStarLink, Asterisk, and more. No SSH client is required although use of SSH directly is fully supported.

    "},{"location":"user-guide/pi-detailed/#quick-config-via-asl-menu","title":"Quick Config via ASL-Menu","text":"

    The node may now be configured. New users or those wanting a quick start experience should use the asl-menu utility with sudo asl-menu.

    The menu will appear:

    "},{"location":"user-guide/pi-detailed/#asterisk-console","title":"Asterisk Console","text":"

    To start the asterisk console from the Terminal window enter sudo asterisk -rv.

    "},{"location":"user-guide/pi-detailed/#software-updates","title":"Software Updates","text":"

    One should also apply any latest updates and reboot the device as described in Cockpit Updates.

    "},{"location":"user-guide/pi-detailed/#allmon3-default-login","title":"Allmon3 Default Login","text":"

    The default-configured username and password combination is allmon3 / password.

    You must delete this and add a new one.

    Details on updating the login(s) and other Allmon3 settings can be found in the Allmon3 section of the manual..

    "},{"location":"user-guide/pi-detailed/#logging-out","title":"Logging Out","text":"

    When finished with the admin system, click on Session in the upper right and then choose Logout.

    "},{"location":"user-guide/pi-detailed/#node-configuration","title":"Node Configuration","text":"

    Next step is to configure the node settings. YouTuber Freddie Mac has a nice ASL3 installation and configuration video. See https://youtu.be/aeuj-yI8qrU. Also see ASL3 Menu for details.

    "},{"location":"user-guide/support/","title":"ASL3 Support","text":""},{"location":"user-guide/support/#support","title":"Support","text":"

    Primary support for all versions of ASL (including ASL3) continues to be the AllStarLink Community. Please do not use the ASL Helpdesk for technical help. Support for ASL3 will NOT be handled by the ASL Helpdesk. All Helpdesk tickets will be closed with a referral to the AllStarLink Community or GitHub.

    "},{"location":"user-guide/support/#bug-reporting","title":"Bug Reporting","text":"

    Bug reports for ASL3 and surrounding issues are best handled through GitHub \"Issues\" to the source code projects at AllStarLink GitHub. There are numerous ASL3 repositories. If you're not sure which repository to use for reporting a given issue then start at ASL3. You can also ask for guidance in the AllStarLink Community.

    When reporting an issue we ask that you be comfortable enough with Linux to edit files, restart services, and perhaps copy a potentially fixed file onto your system.

    Note: to raise issues about ASL3 problems and interact with the developers you must create (or use an existing) GitHub account.

    "},{"location":"user-guide/support/#enhancement-requests","title":"Enhancement Requests","text":"

    Obviously, we cannot promise anything but we do want to hear your ideas. Please file a GitHub \"Issue\" :-)

    "},{"location":"user-guide/support/#contributing","title":"Contributing","text":"

    Have you identified a problem and have a solution? We welcome your help. Please file a GitHub issue with all the details. Even better, create a GitHub \"Pull request\" for us to review.

    "},{"location":"user-guide/troubleshooting/","title":"Basic Troubleshooting Tips","text":"

    When asking for help on AllStarLink Community or in a GitHub Issue here are some tips on how to ask for help and how to get common troubleshooting information that is often needed to understand your specific issues.

    "},{"location":"user-guide/troubleshooting/#tips-on-asking-for-help","title":"Tips on Asking for Help","text":"

    Keep in mind that when asking for help, the pool of people likely able to help have no understanding of your personal setup, situation, skills, etc. In order to get good help, you need to ask a good question. Make sure when asking for help that you hit the following points:

    "},{"location":"user-guide/troubleshooting/#explain-precisely-what-is-not-working-properly","title":"Explain Precisely What is not Working Properly","text":"

    Requests for help such as \"ASL doesn't work\" or \"my pi crashes\" or \"I can't do X\" without any further details or information is unable to be diagnosed and assisted.

    Examples of GOOD ways to ask a question:

    Basically, before submitting your question to Community, make sure your post contains the basics of \"What am I trying to do\", \"what do I think should be happening\", and \"What is happening instead\". If you are filing an issue on GitHub, use the Issue Templates - they ask for information for a reason.

    "},{"location":"user-guide/troubleshooting/#be-ready-to-provide-requested-basic-troubleshooting-info","title":"Be Ready to Provide Requested Basic Troubleshooting Info","text":"

    If you are going to ask for help, be ready and willing to provide the basic troubleshooting information that is requested. For example, if you're having a problem linking to another node, be ready to provide - or even better provide in advance - the basic troubleshooting information needed to help with your problem.

    Items that may be requested include elements such as:

    "},{"location":"user-guide/troubleshooting/#allstarlink-does-not-support-certain-other-projectstools","title":"AllStarLink Does Not Support Certain Other Projects/Tools","text":"

    Core AllStarLink does not support certain other projects/tools such as HamVOIP, Supermon/Supermon2, SkyWarnPlus, or AllScan. Some of the developers of these tools such as SkyWarnPlus and AllScan are active and answer questions on the Community and others are not. If you are told, \"that isn't supported but have you tried doing it in this supported way\", consider giving it a try.

    "},{"location":"user-guide/troubleshooting/#allstarlink-v3-contains-14-years-of-changes","title":"AllStarLink v3 Contains 14+ Years of Changes","text":"

    The release of AllStarLink v3 uses modern Asterisk 20 LTS. Legacy installs use Asterisk 1.4 with was end of life in 2012. Thus, there are literally 14+ years of changes of how Asterisk operates that app_rpt (the main engine behind ASL) had to conform to. Notably, the configuration syntax which changed gradually for time for users staying current with Asterisk is an \"all of a sudden\" change in ASL3. One cannot paste old configuration into modern ASL3 and have it \"just work\".

    Additionally, Asterisk no longer runs as the root user on Linux which means that its ability to do things to the system such as make OS changes is purposefully restricted. ASL3 and Asterisk are very security conscious as many of our users run ASL3 systems on the open Internet.

    "},{"location":"user-guide/troubleshooting/#give-questioners-the-benefit-of-the-doubt","title":"Give Questioners the Benefit of the Doubt","text":"

    In general, if people are asking questions about the details of your problem, they are likely trying to help and not nitpick. In general, if many people are having the problem. it is already known and likely documented. If it's not, then likely your issue is unique to your setup and providing the requested troubleshooting is needed to help.

    "},{"location":"user-guide/troubleshooting/#basic-troubleshooting","title":"Basic Troubleshooting","text":""},{"location":"user-guide/troubleshooting/#gathering-logs","title":"Gathering Logs","text":"

    If you are asked to provide logs, there are two standard ways to provide logs - using Cockput and using the journalctl tool from the command line.

    "},{"location":"user-guide/troubleshooting/#gathering-logs-with-cockpit","title":"Gathering Logs with Cockpit","text":"

    In the Cockpit interface, navigate to Services on the left navbar and click. The service list will display. Wait for the full list to load and then click on the blue link name of the service. The common services are:

    After clicking on one of the links wait until Service logs fills in and then view the logs. Clicking View all logs will show more logs.

    "},{"location":"user-guide/troubleshooting/#gathering-logs-with-clissh","title":"Gathering Logs with CLI/SSH","text":"

    Use the journalctl -xeu command to provide logging output. For example:

    journalctl -xeu asterisk.service\n

    Common services are: * allmon3.service - Allmon 3

    "},{"location":"user-guide/troubleshooting/#gathering-configs","title":"Gathering Configs","text":"

    It is often needed to gather certain configuration details. Use the Cockpit Console or the SSH/CLI interface. People asking for information will likely provide the file needed in the request. However here are some easy ways to get the data requested:

    When possible, please copy/paste text and not screenchots into systems for troubleshooting. Text is far easier to deal with.

    "},{"location":"user-guide/updates/","title":"Updating ASL3","text":"

    All of the software components that make up AllStarLink v3 are provided by software packages (Debian .deb files) from an centralized Apt repository. This makes upgrades quick and painless for users.

    "},{"location":"user-guide/updates/#standard-debian-upgrades","title":"Standard Debian Upgrades","text":"

    Whether AllStarLink v3 is installed on a standard Debian 12 installation or you are using the Pi appliance image, updates follow the standard Debian format:

    sudo apt update\nsudo apt upgrade -y\n

    This will update all AllStarLink v3 software as well as the underlying operating system. AllStarLink will never release any update that is a \"breaking change\" through the apt system that doesn't have either an automated upgrade process or a well-documented manual step-by-step process.

    A reboot is only needed when a new Linux kernel has been installed.

    Sometimes an update may ask what to do with a configuration file. It is generally best to answer such questions with N meaning \"keep your currently-installed version. Such a question will look like:

    Configuration file '/etc/asterisk/rpt.conf'\n ==> Modified (by you or by a script) since installation.\n ==> Package distributor has shipped an updated version.\n   What would you like to do about it ?  Your options are:\n    Y or I  : install the package maintainer's version\n    N or O  : keep your currently-installed version\n      D     : show the differences between the versions\n      Z     : start a shell to examine the situation\n The default action is to keep your current version.\n*** rpt.conf (Y/I/N/O/D/Z) [default=N] ? N\n

    AllStarLink will clearly announce when there is a change that may break existing configurations.

    If you want updates to install without any questions being asked specify the DEBIAN_FRONTEND=noninteractive variable:

    DEBIAN_FRONTEND=noninteractive sudo apt update\nDEBIAN_FRONTEND=noninteractive sudo apt upgrade -y\n

    Using this command will automatically choose \"N\" to any question about configuration - i.e. it will always keep your installed version.

    "},{"location":"user-guide/updates/#note-on-dahdi","title":"Note on DAHDI","text":"

    In the past, managing the DAHDI kernel module needed by app_rpt had been problematic during kernel upgrades. Previously, the recommendation had been to never upgrade the kernel. This advice is no longer valid and should not be followed for security and practicality reasons.

    AllStarLink v3 automatically manages and rebuilds the DAHDI kernel modules during software updates using the DKMS system. After an upgrade, rebooting into a new version of the Linux kernel will \"Just Work\".

    "},{"location":"user-guide/whats-new/","title":"What's New?","text":"

    Ready to get on with installation? We thought so, but first consider that ASL3 is very new. You'll probably want to install on removable media or on a non-production system. You probably don't want to install a production system right away. Give yourself some time to test that everything works as you'd expect.

    Note: if you have installed an earlier version of ASL you need to know that ASL3 has many changes and that there's a bit of a learning curve. For those who have used ASL2 or other versions of AllStar you should be aware of a few important details:

    The two node numbers per user policy remains in effect. Node numbers are a limited resource. Please make a best effort to use new media, a new server, or NNX to build a node for testing. We do make exceptions on a case by case basis but we'd really appreciate it if you could use a resource you already have.

    "},{"location":"user-guide/whats-new/#menu-changes","title":"Menu Changes","text":"

    The ASL3 menu has been updated. Our goal is to provide an easy way for you to configure your node and to minimize the need to edit any configuration files.

    At the same time, we have removed most of the \"OS\" configuration options found in earlier implementations. Historically, ASL has been installed on many OS variants and the OS commands that work on one version do not always work on others. For that reason there are no OS related commands in the ASL3 menu.

    See Menu for details.

    "},{"location":"user-guide/whats-new/#configuration-changes","title":"Configuration Changes","text":"

    The first thing to know about ASL3 configuration is that the menu can be used to make and view common changes. Here are the highlights. You may want to read about Advanced Topics and New Commands details later.

    "},{"location":"user-guide/whats-new/#ap-measurement","title":"A&P Measurement","text":"

    Adoption and Performance measurements are to help ASL understand growth and technical aspects of ASL3. No sensitive or private information is collected. For details see GitHub/AllStarLink/ASL3-Usage. Reports are available at repo.allstarlink.org/reports/.

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