asterisk conf: more configuration
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mpmc
parent
98c4472170
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aeba465e76
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;
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; /etc/asterisk/asterisk.conf
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[directories](!)
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# /etc/systemd/system/asterisk.service
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# "Borrowed" from https://g1fef.co.uk/asterisk-systemd-startup-script/
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[Unit]
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; /etc/asterisk/extensions.conf
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; Remember, context names for each SIP account are specified in pjsip_wizard.conf.
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; First, some safeguards against abuse of the built-in contexts.
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[default]
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exten => _X.,1,Hangup(3)
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[globals]
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; trunk caller IDs.
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TRUNK_ONE_CID = +442012345678
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TRUNK_TWO_CID = +442087456210
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; Default queue
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QUEUE_ONE = queue-one
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[subscribe]
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exten => _X.,hint,PJSIP/${EXTEN}
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[from-trunk-one]
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exten => _X.,1,Queue(${QUEUE_ONE},nr,,,50)
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; same => n,Answer(500)
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; same => n,Voicemail(${VOICEMAIL_BOX},su)
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same => n,Hangup()
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[outbound-one]
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exten => _X.,1,Verbose(1, "OUTBOUND TRUNK ONE ${TRUNK_ONE_CID}")
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same => n,Set(CALLERID(all)=${TRUNK_ONE_CID})
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same => n,Dial(PJSIP/${EXTEN}@trunk-one)
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same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?outbound-two,${EXTEN},1)
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same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?outbound-two,${EXTEN},1)
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same => n,Hangup()
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[outbound-two]
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exten => _X.,1,Verbose(1, "OUTBOUND TRUNK TWO ${TRUNK_TWO_CID}")
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same => n,Set(CALLERID(all)=${TRUNK_TWO_CID})
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same => n,Dial(PJSIP/${EXTEN}@trunk-two)
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same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?default,${EXTEN},1)
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same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?default,${EXTEN},1)
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same => n,Hangup()
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[app-lastcallreturn]
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exten => 1471,1,Wait(1)
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same => n,Answer(1)
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same => n,Playback(last-num-to-call)
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same => n,Set(number=${DB(lastcaller/${CALLERID(num)})})
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same => n,GotoIf($["${number}" = ""]?s-nonum,1)
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same => n,SayDigits(${number})
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same => n,Wait(2)
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same => n,SayDigits(${number})
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same => n,Wait(1)
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same => n,Playback(to-call-num-press)
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same => n,Playback(digits/1)
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exten => 1,1,Goto(outbound-one,${number},1)
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exten => s-nonum,1,Playback(unidentified-no-callback)
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same => n,Hangup()
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[from-ext]
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exten => 123,1,goto(default,${EXTEN},1)
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same => n,Hangup()
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exten => 1471,1,Goto(app-lastcallreturn,${EXTEN},1)
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same => n,Hangup()
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exten => _X.,1,Dial(PJSIP/outbound-one/${EXTEN})
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same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?outbound-two,${EXTEN},1)
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same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?outbound-two,${EXTEN},1)
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same => n,Hangup()
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[to-ext]
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exten => _X.,1,Set(DB(lastcaller/${ARG1})=${CALLERID(num)})
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same => n,Dial(PJSIP/${EXTEN})
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same => n,Hangup()
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@ -1,3 +1,4 @@
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;
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; /etc/asterisk/logger.conf
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;
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; Logging Configuration
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@ -1,3 +1,4 @@
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;
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; /etc/asterisk/modules.conf
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;
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; Asterisk configuration file
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;
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; /etc/asterisk/pjsip.conf
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;
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; Modifed and "borrowed" (read: stolen) from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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; This template contains default settings for all transports.
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[transport-defaults](!)
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type = transport
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bind = 0.0.0.0:15000
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; For communication to any addresses within local_nets, Asterisk will not apply
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; NAT-related workarounds.
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local_net = 127.0.0.0/8
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local_net = 10.0.0.0/8
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local_net = 172.16.0.0/12
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local_net = 192.168.0.0/16
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; If you have a public static IP for your Asterisk server, set it here.
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;external_media_address = xxx.xxx.xxx.xxx
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;external_signaling_address = xxx.xxx.xxx.xxx
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; The following UDP and TCP transports will inherit from the defaults.
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[transport-udp](transport-defaults)
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protocol = udp
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;[transport-tcp](transport-defaults)
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;protocol = tcp
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;
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; /etc/asterisk/pjsip_wizard.conf
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;
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; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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;
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; This template contains default settings for all SIP trunks.
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[trunk-defaults](!)
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type = wizard
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; Send media to the address and port on the incoming packet, regardless of what
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; the SIP headers say (NAT workaround).
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endpoint/rtp_symmetric = yes
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; Rewrite the SIP contact to the address and port of the request (NAT workaround).
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endpoint/rewrite_contact = yes
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; Send the Remote-Party-ID SIP header. Some providers need this.
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endpoint/send_rpid = yes
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; We use alaw mainly in Europe.
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endpoint/allow = !all,alaw
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; Call encryption.
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endpoint/media_encryption = no
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; Load trunks.
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#tryinclude pjsip_wizard_trunk_one.conf
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;
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;
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; This template contains default settings for all local extensions.
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[extension-defaults](!)
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type = wizard
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; Require clients to register.
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accepts_registrations = yes
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; Require clients to authenticate.
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accepts_auth = yes
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; When simultaneous logins from the same account exceed max_contacts, disconnect
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; the oldest session.
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aor/remove_existing = yes
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; allow just alaw
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endpoint/allow = !all,alaw
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; Context name for BLF/presence subscriptions. This can be any string of your
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; choosing.
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endpoint/subscribe_context = subscribe
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; Load user extensions.
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#tryinclude pjsip_wizard_users.conf
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;
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; /etc/asterisk/pjsip_wizard_trunk_one.conf
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;
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; For a local system to register as a trunk.
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;
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[trunk-one](trunk-defaults)
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; Accept inbound auth for this trunk.
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accepts_auth = yes
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; UDP transport please.
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transport = transport-udp
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; What's the username?
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inbound_auth/username = trunk-one
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; The Password?
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inbound_auth/password = BedknobsArentSupposedToWorkThatWayUNLESSYOUREBRAVE
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; Choose a context name for incoming calls from this account. You'll use this
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; name in your dialplan.
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endpoint/context = from-trunk-one
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; Only one registration please.
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aor/max_contacts = 1
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;
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; /etc/asterisk/pjsip_wizard_trunk_two.conf
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;
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; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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;
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[trunk-two](trunk-defaults)
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; Require SIP authentication.
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sends_auth = yes
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; Require SIP registration.
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sends_registrations = yes
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; If registration fails, keep trying until x tries.
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registration/max_retries = 1000
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; Don't assume an authentication failure is permanent.
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registration/auth_rejection_permanent = no
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; Perform a connectivity check every 30 seconds.
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aor/qualify_frequency = 30
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; UDP transport please.
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transport = transport-udp
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; Hostname and port for your SIP provider.
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remote_hosts = voiceless.aa.net.uk:5060
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; Choose a context name for incoming calls from this account. You'll use this
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; name in your dialplan.
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endpoint/context = from-trunk-two
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; Your SIP provider will give you these credentials.
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outbound_auth/username = +442012345678
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outbound_auth/password = NoIDontHaveAnyMONEY
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; /etc/asterisk/pjsip_wizard_users.conf
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;
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; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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;
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[1000](extension-defaults)
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; Dialplan context name for calls originating from this account.
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endpoint/context = from-ext
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; Voicemail address.
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endpoint/mailboxes = 1000@default
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; Internal Caller ID string for this device.
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endpoint/callerid = 1000 <1000>
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; Username for SIP account. By convention, this should be the extension number.
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inbound_auth/username = 1000
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; Password for SIP account (you can choose whatever password you like).
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inbound_auth/password = IfYouTolerateThisThenYourChildrenWillBeNext!
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; Maximum number of simultaneous logins for this account.
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aor/max_contacts = 1
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; Check connectivity every 30 seconds.
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aor/qualify_frequency = 30
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; Set connectivity check timeout to 3 seconds.
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aor/qualify_timeout = 3.0
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; IMPORTANT! This setting determines whether the audio stream will be proxied
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; through the Asterisk server.
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;
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; If this device is directly reachable by the internet (either by a publicly
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; routable IP, or static port mappings on your router), choose YES.
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;
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; Otherwise, if this device is hidden behind NAT, choose NO.
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endpoint/direct_media = no
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; /etc/asterisk/queues.conf
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;
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; Global queue settings go in this section.
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;
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; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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;
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[general]
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; Persist dynamic member lists in the astdb.
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persistentmembers = yes
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; Some options for more intuitive queue behavior.
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autofill = yes
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monitor-type = MixMonitor
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shared_lastcall = yes
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log_membername_as_agent = yes
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;;;;;;;;;;;;;;;;;;;
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; Queue Definitions
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;;;;;;;;;;;;;;;;;;;
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; The first queue is for incoming calls to our lines.
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[queue-one]
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; For each incoming call, ring all members of the queue.
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strategy = ringall
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; Max number of seconds a caller waits in the queue.
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timeout = 90
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; Don't announce estimated hold time, etc.
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announce-frequency = 0
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announce-holdtime = no
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announce-position = no
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periodic-announce-frequency = 0
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; Allow ringing even when no queue members are present.
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joinempty = yes
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leavewhenempty = no
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; Ring member phones even when they are on another call.
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ringinuse = yes
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; Queue Members
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;
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; Each member is specified with the following format:
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; member => INTERFACE,PENALTY,FRIENDLY_NAME,PRESENCE_INTERFACE
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;
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; The "penalty" value is not interesting for our use case.
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; With PJSIP, the BLF/Presence interface is identical to the standard interface name.
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member => to-ext/1000,0,1000,to-ext/1000
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;
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; /etc/asterisk/rtp.conf
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;
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[general]
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stunaddr = stun.aa.net.uk:3478
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stunrefresh = 30
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;
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; /etc/asterisk/rtp.conf
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;
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; Modifed and "borrowed" (read: stolen) from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
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;
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[general]
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rtpstart=15010
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rtpend=15999
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@ -0,0 +1,47 @@
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[general]
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format = wav49|gsm|wav
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[zonemessages]
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; Users may be located in different timezones, or may have different
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; message announcements for their introductory message when they enter
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; the voicemail system. Set the message and the timezone each user
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; hears here. Set the user into one of these zones with the tz= attribute
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; in the options field of the mailbox. Of course, language substitution
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; still applies here so you may have several directory trees that have
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; alternate language choices.
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;
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; Look in /usr/share/zoneinfo/ for names of timezones.
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; Look at the manual page for strftime for a quick tutorial on how the
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; variable substitution is done on the values below.
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;
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; Supported values:
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; 'filename' filename of a soundfile (single ticks around the filename
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; required)
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; ${VAR} variable substitution
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; A or a Day of week (Saturday, Sunday, ...)
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; B or b or h Month name (January, February, ...)
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; d or e numeric day of month (first, second, ..., thirty-first)
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; Y Year
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; I or l Hour, 12 hour clock
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; H Hour, 24 hour clock (single digit hours preceded by "oh")
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; k Hour, 24 hour clock (single digit hours NOT preceded by "oh")
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; M Minute, with 00 pronounced as "o'clock"
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; N Minute, with 00 pronounced as "hundred" (US military time)
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; P or p AM or PM
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; Q "today", "yesterday" or ABdY
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; (*note: not standard strftime value)
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; q "" (for today), "yesterday", weekday, or ABdY
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; (*note: not standard strftime value)
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; R 24 hour time, including minute
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;
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eastern=America/New_York|'vm-received' Q 'digits/at' IMp
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central=America/Chicago|'vm-received' Q 'digits/at' IMp
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central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
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military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
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european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
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uk=Europe/London|'vm-received' a d b 'digits/at' HM
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[default]
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; Note: The rest of the system must reference mailboxes defined here as mailbox@default.
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1000 => 1000,Default Mailbox,,,,Tz=uk
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