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asterisk

Website

Logging in

Asterisk is on watson.fsf.org server.

ssh root@watson.fsf.org

Debugging

If there is choppiness on the conference bridge (meetme), login to watson.fsf.org and become root, then try the following:

Pull up the asterisk console

asterisk -vr

Look to see if there are ghosts or duplicate users on the bridge:

 asterisk> confbridge list 7600

If you do not see anyone, they might be in a different room. You can use the command from Check for users before restarting to list all participants.

This should show you something like the following:

Channel                        Flags  User Profile     Bridge Profile   Menu             CallerID
============================== ====== ================ ================ ================ ================
SIP/jrasata-00000015                  default_user     default_bridge   default_menu     jrasata

Old instructions: Notice that User #: 02 is the same as #08. In this case we can tell that the channel SIP/johns-00000083 has been open for 5 hours and 27 mins, this is obvious a ghost and causing the meetme plugin to generate additional traffic that the client cannot receive (causing choppy sounds over the bridge).

The solution here is to kill the channel, meetme has a clean way of doing this from the plugin itself with the following command:

asterisk> confbridge kick 7600 SIP/jrasata-00000015

This command kicks the User from the bridge and kills its channel cleanly. At this point the conference bridge should be back to normal.

When you are done, keep the asterisk shell up and running in a tmux session. This displays more useful recent info than the regular file-based logs do.

Check for users before restarting

asterisk> core show channels

fail2ban and firewall

If you restore the firewall, ensure that you restart the fail2ban service, or our host will be unprotected from brute force attempts and this will also reduce the performance of the asterisk server. To verify correctness, look for fail2ban sections in the output of "iptables -L -v"

CLI-help

When you are in the asterisk CLI, you can type ? or TAB at any point in your command for suggested command completions

Reloading modules

After editing extensions.conf, run the following in the asterisks CLI:

> dialplan reload

After editing sip.conf:

> sip reload

If that is not enough (this does more than systemctl reload):

> reload

Files you will likely edit

Under /etc/asterisk/:

extensions.conf (dialplan)
sip.conf (SIP config, to be replaced by pjsip.conf when we are ready)
voicemail.conf (voicemail)
confbridge.conf (conference calling)
chan_dahdi.conf (for editing physical phone lines)

Extensions.conf

First of all, read the README-SERIOUSLY.bestpractices.md file linked below. We must not allow others to commit toll fraud (dialing arbitrary numbers from external calls) with our phone system, otherwise we will face a large bill and/or downtime. There is also a risk of arbitrary code execution if we are not careful.

Security highlights:

Use strict pattern matching, avoid using . or !. use FILTER() to avoid injection from arbitrary user input via caller id or dialed strings.
Do not use numbers for SIP device names (can be brute forced).
Secure passwords.
Avoid typos by using same => n,... syntax.
Avoid use of EXEC* to avoid arbitrary code execution.
Keep live_dangerously set to no.

Extension number formatting (from PDF linked below):

_ denotes a pattern matching extension
N matches any number from 2 through 9
X matches any single digit
. matches one or more of any digit
[2-6] matches any of 2,3,4,5,6

Call Andrew according to ANDREW=SIP/andrew/ variable set above:

#        extension,rule_number,command
exten => 39,1,NoOp(About to call Andrew's SIP line) # prints to console
exten => 39,2,Dial(${ANDREW},5,t)

Do more tasks based on order of inclusion, rather than explicit numbering

exten => 39,n,NoOp(Done calling Andrew's SIP line)
exten => 39,n,NoOp(Done calling Andrew's SIP line)
same  =>    n,NoOp(Done calling Andrew's SIP line)

(I am not sure if those commands get processed if they reach voicemail, but a hangup event from either party should terminate the rule processing).

The s extension stands for a special (or start?) extension

exten =>  0,1,GoTo(4,7)
exten =>  4,7,GoTo(context-name,s,1)

[context-name]
exten =>  s,1,NoOp(Got to this rule)

Note that parameters in () in at least some cases are required / counted from right to left. So you can leave off the beginning of a parameter list to be more concise, but you must include parameters through the end.

There is extensive documentation in the extensions.conf file itself, but you will have to scroll down to the appropriate section to find it. Try searching the file for Any category other.

Do not forget to reload the Asterisk configs after making changes.

sip.conf

We use allowguest=yes in our sip.conf file. This allows other SIP hosts to connect to our SIP server without a user account password, and without us allowlisting their IP address. Some companies might not want that, as they want to be dialed by a phone number via the SIP trunk of their SIP provider, and none of their customers use SIP to connect directly to other hosts. In our case, if Free Software Org Foo wants to connect from their SIP server to ours, disallowing guests would not allow them to connect.

https://kb.smartvox.co.uk/asterisk/secure-asterisk-pbx-part-2/

Custom voicemail name / greeting

For RMS' mailbox 43, the automated voicemail voice replies splices RMS' name in his own voice, rather than saying "extension 43":

/var/spool/asterisk/voicemail/default/43/greet.gsm

file -sL has this to say about the above file:

greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz

So if someone wants the voicemail to refer to them by name, ask them to record their name into a microphone, then convert it into a similar format as above.

Call logs

Call logs are found in the /var/log/asterisk/cdr-csv/Master.csv file where cdr stands for Call Detail Records.

An outgoing call starts with a sip account and is followed by the number that they dialed like so:

"","michael","918657127946",

An incoming call starts with an external number and is followed by the extension they used like so:

"","18657127946","501",

Verbose, recent logs

The best way to see detailed logs is to see them on the console of asterisk -vvvvr, because not all messages are logged to files, but they do show at that command prompt. For this reason, I like to keep a tmux session open with this command, so we can look at recent lines if any issue comes up.

Random notes

& means to ring multiple phones at the same time.

We have billing limits with callcentric, they will pause our account if we spend too much money, for example if someone figures out how to do fraud through our asterisk.

There are protocols for t.38, which should allow us to send faxes, but not sure if our fax machine supports it.