diff --git a/asterisk/asterisk.conf b/asterisk/asterisk.conf
index 2723ac9..0542c93 100644
--- a/asterisk/asterisk.conf
+++ b/asterisk/asterisk.conf
@@ -1,3 +1,4 @@
+;
 ; /etc/asterisk/asterisk.conf
 
 [directories](!)
diff --git a/asterisk/asterisk.service b/asterisk/asterisk.service
index 6400d3e..e05cd96 100644
--- a/asterisk/asterisk.service
+++ b/asterisk/asterisk.service
@@ -1,3 +1,5 @@
+# /etc/systemd/system/asterisk.service
+
 # "Borrowed" from https://g1fef.co.uk/asterisk-systemd-startup-script/
 
 [Unit]
diff --git a/asterisk/extensions.conf b/asterisk/extensions.conf
new file mode 100644
index 0000000..aa53c34
--- /dev/null
+++ b/asterisk/extensions.conf
@@ -0,0 +1,79 @@
+; /etc/asterisk/extensions.conf
+
+; Remember, context names for each SIP account are specified in pjsip_wizard.conf.
+
+; First, some safeguards against abuse of the built-in contexts.
+[default]
+exten => _X.,1,Hangup(3)
+
+[globals]
+
+; trunk caller IDs.
+TRUNK_ONE_CID = +442012345678
+TRUNK_TWO_CID = +442087456210
+
+; Default queue
+QUEUE_ONE = queue-one
+
+[subscribe]
+exten => _X.,hint,PJSIP/${EXTEN}
+
+[from-trunk-one]
+exten => _X.,1,Queue(${QUEUE_ONE},nr,,,50)
+; same => n,Answer(500)
+; same => n,Voicemail(${VOICEMAIL_BOX},su)
+ same => n,Hangup()
+
+
+[outbound-one]
+exten => _X.,1,Verbose(1, "OUTBOUND TRUNK ONE ${TRUNK_ONE_CID}")
+ same => n,Set(CALLERID(all)=${TRUNK_ONE_CID})
+ same => n,Dial(PJSIP/${EXTEN}@trunk-one)
+ same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?outbound-two,${EXTEN},1)
+ same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?outbound-two,${EXTEN},1)
+ same => n,Hangup()
+
+[outbound-two]
+exten => _X.,1,Verbose(1, "OUTBOUND TRUNK TWO ${TRUNK_TWO_CID}")
+ same => n,Set(CALLERID(all)=${TRUNK_TWO_CID})
+ same => n,Dial(PJSIP/${EXTEN}@trunk-two)
+ same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?default,${EXTEN},1)
+ same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?default,${EXTEN},1)
+ same => n,Hangup()
+
+[app-lastcallreturn]
+exten => 1471,1,Wait(1)
+ same => n,Answer(1)
+ same => n,Playback(last-num-to-call)
+ same => n,Set(number=${DB(lastcaller/${CALLERID(num)})})
+ same => n,GotoIf($["${number}" = ""]?s-nonum,1)
+ same => n,SayDigits(${number})
+ same => n,Wait(2)
+ same => n,SayDigits(${number})
+ same => n,Wait(1)
+ same => n,Playback(to-call-num-press)
+ same => n,Playback(digits/1)
+
+exten => 1,1,Goto(outbound-one,${number},1)
+
+exten => s-nonum,1,Playback(unidentified-no-callback)
+ same => n,Hangup()
+
+
+[from-ext]
+exten => 123,1,goto(default,${EXTEN},1)
+ same => n,Hangup()
+
+exten => 1471,1,Goto(app-lastcallreturn,${EXTEN},1)
+ same => n,Hangup()
+
+exten => _X.,1,Dial(PJSIP/outbound-one/${EXTEN})
+ same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?outbound-two,${EXTEN},1)
+ same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?outbound-two,${EXTEN},1)
+ same => n,Hangup()
+
+[to-ext]
+exten => _X.,1,Set(DB(lastcaller/${ARG1})=${CALLERID(num)})
+ same => n,Dial(PJSIP/${EXTEN})
+ same => n,Hangup()
+
diff --git a/asterisk/logger.conf b/asterisk/logger.conf
index a708f0f..5f8dcb4 100644
--- a/asterisk/logger.conf
+++ b/asterisk/logger.conf
@@ -1,3 +1,4 @@
+;
 ; /etc/asterisk/logger.conf
 ;
 ; Logging Configuration
diff --git a/asterisk/modules.conf b/asterisk/modules.conf
index 0a3b82f..76bc9c7 100644
--- a/asterisk/modules.conf
+++ b/asterisk/modules.conf
@@ -1,3 +1,4 @@
+;
 ; /etc/asterisk/modules.conf
 ;
 ; Asterisk configuration file
diff --git a/asterisk/pjsip.conf b/asterisk/pjsip.conf
new file mode 100644
index 0000000..0d795ee
--- /dev/null
+++ b/asterisk/pjsip.conf
@@ -0,0 +1,27 @@
+;
+; /etc/asterisk/pjsip.conf
+;
+; Modifed and "borrowed" (read: stolen) from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
+
+; This template contains default settings for all transports. 
+[transport-defaults](!)
+type = transport
+bind = 0.0.0.0:15000
+
+; For communication to any addresses within local_nets, Asterisk will not apply
+; NAT-related workarounds.
+local_net = 127.0.0.0/8
+local_net = 10.0.0.0/8
+local_net = 172.16.0.0/12
+local_net = 192.168.0.0/16
+
+; If you have a public static IP for your Asterisk server, set it here.
+;external_media_address     = xxx.xxx.xxx.xxx
+;external_signaling_address = xxx.xxx.xxx.xxx
+
+; The following UDP and TCP transports will inherit from the defaults.
+[transport-udp](transport-defaults)
+protocol = udp
+
+;[transport-tcp](transport-defaults)
+;protocol = tcp
diff --git a/asterisk/pjsip_wizard.conf b/asterisk/pjsip_wizard.conf
new file mode 100644
index 0000000..f1ea9ab
--- /dev/null
+++ b/asterisk/pjsip_wizard.conf
@@ -0,0 +1,56 @@
+;
+; /etc/asterisk/pjsip_wizard.conf
+;
+; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
+;
+; This template contains default settings for all SIP trunks.
+[trunk-defaults](!)
+type = wizard
+
+; Send media to the address and port on the incoming packet, regardless of what
+; the SIP headers say (NAT workaround).
+endpoint/rtp_symmetric = yes
+
+; Rewrite the SIP contact to the address and port of the request (NAT workaround).
+endpoint/rewrite_contact = yes
+
+; Send the Remote-Party-ID SIP header. Some providers need this.
+endpoint/send_rpid = yes
+
+; We use alaw mainly in Europe.
+endpoint/allow = !all,alaw
+
+; Call encryption.
+endpoint/media_encryption = no
+
+; Load trunks.
+#tryinclude pjsip_wizard_trunk_one.conf
+;
+
+;
+; This template contains default settings for all local extensions.
+[extension-defaults](!)
+type = wizard
+
+; Require clients to register.
+accepts_registrations = yes
+
+; Require clients to authenticate.
+accepts_auth = yes
+
+; When simultaneous logins from the same account exceed max_contacts, disconnect
+; the oldest session.
+aor/remove_existing = yes
+
+; allow just alaw
+endpoint/allow = !all,alaw
+
+; Context name for BLF/presence subscriptions. This can be any string of your
+; choosing.
+endpoint/subscribe_context = subscribe
+
+; Load user extensions. 
+#tryinclude pjsip_wizard_users.conf
+
+
+
diff --git a/asterisk/pjsip_wizard_trunk_one.conf b/asterisk/pjsip_wizard_trunk_one.conf
new file mode 100644
index 0000000..3f430f9
--- /dev/null
+++ b/asterisk/pjsip_wizard_trunk_one.conf
@@ -0,0 +1,24 @@
+;
+; /etc/asterisk/pjsip_wizard_trunk_one.conf
+;
+; For a local system to register as a trunk.
+;
+[trunk-one](trunk-defaults)
+; Accept inbound auth for this trunk.
+accepts_auth = yes
+
+; UDP transport please.
+transport = transport-udp
+
+; What's the username?
+inbound_auth/username = trunk-one
+
+; The Password?
+inbound_auth/password = BedknobsArentSupposedToWorkThatWayUNLESSYOUREBRAVE
+
+; Choose a context name for incoming calls from this account. You'll use this
+; name in your dialplan.
+endpoint/context = from-trunk-one
+
+; Only one registration please.
+aor/max_contacts = 1
diff --git a/asterisk/pjsip_wizard_trunk_two.conf b/asterisk/pjsip_wizard_trunk_two.conf
new file mode 100644
index 0000000..85abd82
--- /dev/null
+++ b/asterisk/pjsip_wizard_trunk_two.conf
@@ -0,0 +1,34 @@
+;
+; /etc/asterisk/pjsip_wizard_trunk_two.conf
+;
+; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
+;
+[trunk-two](trunk-defaults)
+; Require SIP authentication.
+sends_auth = yes
+
+; Require SIP registration.
+sends_registrations = yes
+
+; If registration fails, keep trying until x tries.
+registration/max_retries = 1000
+
+; Don't assume an authentication failure is permanent.
+registration/auth_rejection_permanent = no
+
+; Perform a connectivity check every 30 seconds.
+aor/qualify_frequency = 30
+
+; UDP transport please.
+transport = transport-udp
+
+; Hostname and port for your SIP provider.
+remote_hosts = voiceless.aa.net.uk:5060
+
+; Choose a context name for incoming calls from this account. You'll use this
+; name in your dialplan.
+endpoint/context = from-trunk-two
+
+; Your SIP provider will give you these credentials.
+outbound_auth/username = +442012345678
+outbound_auth/password = NoIDontHaveAnyMONEY
diff --git a/asterisk/pjsip_wizard_users.conf b/asterisk/pjsip_wizard_users.conf
new file mode 100644
index 0000000..a269d73
--- /dev/null
+++ b/asterisk/pjsip_wizard_users.conf
@@ -0,0 +1,37 @@
+; /etc/asterisk/pjsip_wizard_users.conf
+;
+; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
+;
+[1000](extension-defaults)
+; Dialplan context name for calls originating from this account.
+endpoint/context = from-ext
+
+; Voicemail address.
+endpoint/mailboxes = 1000@default
+
+; Internal Caller ID string for this device.
+endpoint/callerid = 1000 <1000>
+
+; Username for SIP account. By convention, this should be the extension number.
+inbound_auth/username = 1000
+
+; Password for SIP account (you can choose whatever password you like).
+inbound_auth/password = IfYouTolerateThisThenYourChildrenWillBeNext!
+
+; Maximum number of simultaneous logins for this account.
+aor/max_contacts = 1
+
+; Check connectivity every 30 seconds.
+aor/qualify_frequency = 30
+
+; Set connectivity check timeout to 3 seconds.
+aor/qualify_timeout = 3.0
+
+; IMPORTANT! This setting determines whether the audio stream will be proxied
+; through the Asterisk server.
+;
+; If this device is directly reachable by the internet (either by a publicly
+; routable IP, or static port mappings on your router), choose YES.
+;
+; Otherwise, if this device is hidden behind NAT, choose NO.
+endpoint/direct_media = no
diff --git a/asterisk/queues.conf b/asterisk/queues.conf
new file mode 100644
index 0000000..629e59c
--- /dev/null
+++ b/asterisk/queues.conf
@@ -0,0 +1,49 @@
+; /etc/asterisk/queues.conf
+;
+; Global queue settings go in this section.
+;
+; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
+;
+[general]
+; Persist dynamic member lists in the astdb.
+persistentmembers = yes
+
+; Some options for more intuitive queue behavior.
+autofill                = yes
+monitor-type            = MixMonitor
+shared_lastcall         = yes
+log_membername_as_agent = yes
+
+;;;;;;;;;;;;;;;;;;;
+; Queue Definitions
+;;;;;;;;;;;;;;;;;;;
+
+; The first queue is for incoming calls to our lines.
+[queue-one]
+; For each incoming call, ring all members of the queue.
+strategy = ringall
+
+; Max number of seconds a caller waits in the queue.
+timeout = 90
+
+; Don't announce estimated hold time, etc.
+announce-frequency          = 0
+announce-holdtime           = no
+announce-position           = no
+periodic-announce-frequency = 0
+
+; Allow ringing even when no queue members are present.
+joinempty      = yes
+leavewhenempty = no
+
+; Ring member phones even when they are on another call.
+ringinuse = yes
+
+; Queue Members
+;
+; Each member is specified with the following format:
+;  member => INTERFACE,PENALTY,FRIENDLY_NAME,PRESENCE_INTERFACE
+;
+; The "penalty" value is not interesting for our use case.
+; With PJSIP, the BLF/Presence interface is identical to the standard interface name.
+member => to-ext/1000,0,1000,to-ext/1000
diff --git a/asterisk/res_stun_monitor.conf b/asterisk/res_stun_monitor.conf
new file mode 100644
index 0000000..6ce21df
--- /dev/null
+++ b/asterisk/res_stun_monitor.conf
@@ -0,0 +1,7 @@
+;
+; /etc/asterisk/rtp.conf
+;
+[general]
+
+stunaddr = stun.aa.net.uk:3478
+stunrefresh = 30
diff --git a/asterisk/rtp.conf b/asterisk/rtp.conf
new file mode 100644
index 0000000..07d4938
--- /dev/null
+++ b/asterisk/rtp.conf
@@ -0,0 +1,8 @@
+;
+; /etc/asterisk/rtp.conf
+;
+; Modifed and "borrowed" (read: stolen) from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
+;
+[general]
+rtpstart=15010
+rtpend=15999
diff --git a/asterisk/voicemail.conf b/asterisk/voicemail.conf
new file mode 100644
index 0000000..59abc8e
--- /dev/null
+++ b/asterisk/voicemail.conf
@@ -0,0 +1,47 @@
+[general]
+format = wav49|gsm|wav
+
+[zonemessages]
+; Users may be located in different timezones, or may have different
+; message announcements for their introductory message when they enter
+; the voicemail system. Set the message and the timezone each user
+; hears here. Set the user into one of these zones with the tz= attribute
+; in the options field of the mailbox. Of course, language substitution
+; still applies here so you may have several directory trees that have
+; alternate language choices.
+;
+; Look in /usr/share/zoneinfo/ for names of timezones.
+; Look at the manual page for strftime for a quick tutorial on how the
+; variable substitution is done on the values below.
+;
+; Supported values:
+; 'filename'    filename of a soundfile (single ticks around the filename
+;               required)
+; ${VAR}        variable substitution
+; A or a        Day of week (Saturday, Sunday, ...)
+; B or b or h   Month name (January, February, ...)
+; d or e        numeric day of month (first, second, ..., thirty-first)
+; Y             Year
+; I or l        Hour, 12 hour clock
+; H             Hour, 24 hour clock (single digit hours preceded by "oh")
+; k             Hour, 24 hour clock (single digit hours NOT preceded by "oh")
+; M             Minute, with 00 pronounced as "o'clock"
+; N             Minute, with 00 pronounced as "hundred" (US military time)
+; P or p        AM or PM
+; Q             "today", "yesterday" or ABdY
+;               (*note: not standard strftime value)
+; q             "" (for today), "yesterday", weekday, or ABdY
+;               (*note: not standard strftime value)
+; R             24 hour time, including minute
+;
+eastern=America/New_York|'vm-received' Q 'digits/at' IMp
+central=America/Chicago|'vm-received' Q 'digits/at' IMp
+central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
+military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
+european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
+uk=Europe/London|'vm-received' a d b 'digits/at' HM
+
+[default]
+; Note: The rest of the system must reference mailboxes defined here as mailbox@default.
+
+1000 => 1000,Default Mailbox,,,,Tz=uk