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--- notes:asterisk [2022/06/03 16:26] – maff
+++ notes:asterisk [2022/06/06 10:28] (current) – maff
@@ Line 124: / Line 124: @@
 This part was initially extremely daunting. Not helping matters is that the fact that there are no pages online (that I could find) that discuss a viable minimal configuration for a recent Asterisk version. Apparently either nobody is starting fresh with Asterisk in 2022 or nobody is interested in writing about it. Further frustrating matters is that Asterisk's official wiki, a Confluence site, went down shortly after I began the process of setting Asterisk up.
-===== modules.conf =====
+===== Files removed =====
-I started here, and essentially trimmed the config file down to ''autoload=yes'', because I disabled all modules in ''make menuconfig'' that I didn't want to actually use.
+I deleted the config files installed by ''make samples'' that were installed for all modules I explicitly removed:
-<code ini modules.conf>
-[modules]
-autoload=yes
-</code>
-===== features.conf =====
-I then went here, and I trimmed it down to just the things that seemed relevant.
-<code ini features.conf>
-[general]
-transferdigittimeout	=> 3
-xfersound		= beep
-xferfailsound		= beeperr
-;pickupsound		= beep
-;pickupfailsound	= beeperr
-featuredigittimeout	= 1000
-;recordingfailsound	= beeperr
-atxfernoanswertimeout	= 15
-atxferdropcall		= no
-atxferloopdelay		= 10
-atxfercallbackretries	= 2
-transferdialattempts	= 3
-transferretrysound	= beep
-transferinvalidsound	= beeperr
-atxferabort		= *1
-atxfercomplete		= *2
-atxferthreeway		= *3
-atxferswap		= *4
-pickupexten		= *8
-[featuremap]
-; requires two channels to be both answered and bridged, chan_local is needed w/ Answer in order to use them while RP is ringing or in progress
-disconnect		=> *0 ;requires H or h
-automon			=> *1 ;requires W or w
-atxfer			=> *2 ;requires T or t
-automixmon		=> *3 ;requires X or x
-blindxfer		=> #1 ;requires T or t
-parkcall		=> #72;requires X or x
-[applicationmap]
-</code>
-===== Config files removed or just trimmed down inconsequentially =====
-This section then is for config files I edited or deleted from what is installed with ''make samples''.
-I deleted the config files that were installed for all modules I explicitly removed:
   * app_skel.conf
@@ Line 198: / Line 148: @@
   * vpb.conf
-And edited down the following:
+===== Files edited =====
+<code ini acl.conf>
+[acl_deny_default]
+deny    = 0.0.0.0/0
+deny    = ::
+[acl_permit_default]
+permit  = 0.0.0.0/0
+permit  = ::
+[acl_local_subnets]
+permit  = 10.13.37.0/24
+permit  = 10.46.0.0/16
+permit  = 172.31.255.0/28
+permit  = fd46::/16
+[acl_permit_local_only]
+deny    = 0.0.0.0/0
+deny    = ::
+permit  = 10.13.37.0/24
+permit  = 10.46.0.0/16
+permit  = 172.31.255.0/28
+permit  = fd46::/16
+</code>
+<code ini adsi.conf>
+[intro]
+alignment = center
+greeting => hewwo
+</code>
+<code ini cli.conf>
+[startup_commands]
+;sip set debug on       = yes
+;core set verbose 3     = yes
+;core set debug 1       = yes
+</code>
 <code ini codecs.conf>
@@ Line 273: / Line 260: @@
 pp_dereverb_decay			=> 0.4
 pp_dereverb_level			=> 0.3
+</code>
+<code ini console.conf>
+[general]
+[default]
+active = no
 </code>
 <code ini extconfig.conf>
 [settings]
-; file.conf => driver,database[,table[,priority]]
+</code>
-;queues.conf => odbc,asterisk,ast_config
-;extensions.conf => sqlite,asterisk,ast_config
+<code ini features.conf>
-;example => odbc,asterisk,alttable,1
+[general]
-;example => mysql,asterisk,alttable,2
+transferdigittimeout	=> 3
-;example2 => ldap,"dc=oxymium,dc=net",example2
+xfersound		= beep
-;iaxusers => odbc,asterisk
+xferfailsound		= beeperr
-;iaxpeers => odbc,asterisk
+;pickupsound		= beep
-;sippeers => odbc,asterisk
+;pickupfailsound	= beeperr
-;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
+featuredigittimeout	= 1000
-;ps_endpoints => odbc,asterisk
+;recordingfailsound	= beeperr
-;ps_auths => odbc,asterisk
+atxfernoanswertimeout	= 15
-;ps_aors => odbc,asterisk
+atxferdropcall		= no
-;ps_domain_aliases => odbc,asterisk
+atxferloopdelay		= 10
-;ps_endpoint_id_ips => odbc,asterisk
+atxfercallbackretries	= 2
-;ps_outbound_publishes => odbc,asterisk
+transferdialattempts	= 3
-;ps_inbound_publications = odbc,asterisk
+transferretrysound	= beep
-;ps_asterisk_publications = odbc,asterisk
+transferinvalidsound	= beeperr
-;voicemail => odbc,asterisk
-;extensions => odbc,asterisk
+atxferabort		= *1
-;meetme => mysql,general
+atxfercomplete		= *2
-;queues => odbc,asterisk
+atxferthreeway		= *3
-;queue_members => odbc,asterisk
+atxferswap		= *4
-;queue_rules => odbc,asterisk
+pickupexten		= *8
-;acls => odbc,asterisk
-;musiconhold => mysql,general
+[featuremap]
-;musiconhold_entry => mysql,general
+; requires two channels to be both answered and bridged, chan_local is needed w/ Answer in order to use them while RP is ringing or in progress
-;queue_log => mysql,general
+disconnect		=> *0 ;requires H or h
+automon			=> *1 ;requires W or w
+atxfer			=> *2 ;requires T or t
+automixmon		=> *3 ;requires X or x
+blindxfer		=> #1 ;requires T or t
+parkcall		=> #72;requires X or x
+[applicationmap]
 </code>
@@ Line 314: / Line 315: @@
 ;capture_id = 1234
 ;uuid_type = call-id | channel
+</code>
+<code ini http.conf>
+[general]
+servername		= Puppybarks
+enabled			= yes
+bindaddr		= 0.0.0.0
+bindport		= 8088
+tlsenable		= no
+;prefix=asterisk
+;sessionlimit=100
+;session_inactivity=30000
+;session_keep_alive=15000
+;enable_static=yes
+;enable_status=no
+;redirect = / /static/config/index.html
+;tlsenable=yes
+;tlsbindaddr=0.0.0.0:8089
+;
+;tlscertfile=
+;tlsprivatekey=
+; tlscipher=
+; ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:
+; ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:
+; kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:
+; ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:
+; ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:
+; DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:
+; AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:
+; AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:
+; !EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA
+; tlsdisablev1=yes
+; tlsdisablev11=yes
+; tlsdisablev12=yes
+; tlsservercipherorder=yes
+;[post_mappings]
+;uploads = /var/lib/asterisk/uploads/
+</code>
+<code ini modules.conf>
+[modules]
+autoload=yes
 </code>
@@ Line 322: / Line 365: @@
 uri                     = metrics
 </code>
+===== pjsip.conf =====
+It was at this point that I gave up because the file is approximately three bibles long and simultaneously contains every example known to humanity and yet no information I'm capable of parsing and retaining. I'll come back to this later.
+====== Dialplan notes ======
+===== Trunk configuration =====
+A&A SIP trunks don't seem to indicate the incoming number, or maybe I configured the trunks on A&A's end wrongly. The solution is simply to specify a contact header, so I have devised the following internal dialplan for inbound trunks
+44 1382 00 339
+Where 8 is the incoming number prefix, 44 is the country-code, 1382 is the area code of the line, 00 is the provider number, and 339 is the last few digits.
+Provider numbers are as follows
+- A&A VoIP
+- Voipfone
+- sipgate
+- Twilio
+- SIPBroker

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