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ast: spacing & trunk defaults

This commit is contained in:
mpmc 2023-07-04 13:10:55 +01:00
parent 073776f22c
commit 01ae1f0830
8 changed files with 98 additions and 64 deletions
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8
asterisk/conf/mobile/extensions.conf
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@ -4,11 +4,11 @@
;
; First, some safeguards against abuse of the built-in contexts.
[default]
exten=>s,1,Verbose(1, "INFO ${CALLERID(num)} was passed to default.")
same=>n,Hangup(3)
exten => s,1,Verbose(1, "INFO ${CALLERID(num)} was passed to default.")
same => n,Hangup(3)
[presence]
exten=>_X.,hint,PJSIP/${EXTEN}
exten => _X.,hint,PJSIP/${EXTEN}
[from-ext]
exten=>123,1,Hangup(3)
exten => 123,1,Hangup(3)

2
asterisk/conf/mobile/modules.conf
View File

@ -4,7 +4,7 @@
; This is the same as the sample but without text.
;
[modules]
autoload=yes
autoload = yes
noload = chan_alsa.so
noload = res_hep.so

7
asterisk/conf/mobile/pjsip.conf
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@ -11,7 +11,7 @@ type = global
endpoint_identifier_order = ip,username,anonymous
; Basic UDP Transport.
[ipv4-udp-nat]
[ipv4-udp]
type = transport
bind = 0.0.0.0:65534
protocol = udp
@ -19,8 +19,3 @@ local_net = 172.16.0.0/12
local_net = 192.168.0.0/16
local_net = 127.0.0.0/8
local_net = 10.0.0.0/8
;
; Load trunk identify block.
#tryinclude pjsip_identify_trunk_one.conf

32
asterisk/conf/mobile/pjsip_users.conf
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@ -2,11 +2,35 @@
; /etc/asterisk/pjsip_users.conf
;
[1000](extension-defaults)
; Dialplan context name for calls originating from this account.
endpoint/context = from-ext
inbound_auth/username=1000
; Voicemail address.
endpoint/mailboxes = 1000@default
inbound_auth/password=MapleCarrotBlueGrainFishSoapSoup
; Internal Caller ID string for this device.
endpoint/callerid = 1000 <1000>
endpoint/callerid=1000 <Ext 1000>
; Username for SIP account. By convention, this should be the extension number.
inbound_auth/username = 1000
endpoint/context=from-ext
; Password for SIP account (you can choose whatever password you like).
inbound_auth/password = MapleCarrotBlueGrainFishSoapSoup
; 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

44
asterisk/conf/mobile/pjsip_wizard.conf
View File

@ -1,36 +1,60 @@
;
; /etc/asterisk/pjsip_wizard.conf
;
[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
; Default language, pjsip wizard appears to ignore global setting.
endpoint/language = en_GB
;
[extension-defaults](!)
type=wizard
type = wizard
; Users must register and auth!
accepts_registrations=yes
accepts_auth=yes
accepts_registrations = yes
accepts_auth = yes
; Check ext connectivity every x seconds.
aor/qualify_frequency=30
aor/qualify_frequency = 30
; How long do we wait for a response to our connection check?
aor/qualify_timeout=3.0
aor/qualify_timeout = 3.0
; Remove older sessions when logins exceed max_contacts.
aor/remove_existing=yes
aor/remove_existing = yes
; Max simultaneous account logins.
endpoint/max_contacts=1
endpoint/max_contacts = 1
; Audio codecs: alaw only please.
endpoint/allow=!all,alaw
; Default language.
endpoint/language=en_GB
endpoint/language = en_GB
; By default don't allow direct media connections.
endpoint/direct_media=no
endpoint/direct_media = no
; Busy Lamp field / Presence context.
endpoint/subscribe_context=presence
endpoint/subscribe_context = presence
;
; Load user extensions.

61
asterisk/conf/mobile/quectel.conf
View File

@ -1,8 +1,8 @@
[general]
interval=30 ; Number of seconds between trying to connect to devices
smsdb=/var/lib/asterisk/smsdb
csmsttl=600
interval = 30 ; Number of seconds between trying to connect to devices
smsdb = /var/lib/asterisk/smsdb
csmsttl = 600
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
@ -41,35 +41,35 @@ csmsttl=600
; now you can set here any not required device settings as template
; sure you can overwrite in any [device] section this default values
context=default ; context for incoming calls
group=0 ; calling group
rxgain=0 ; increase the incoming volume; may be negative
txgain=0 ; increase the outgoint volume; may be negative
autodeletesms=yes ; auto delete incoming sms
resetquectel=yes ; reset quectel during initialization with ATZ command
u2diag=-1 ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=yes ; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation by default use default network settings
disablesms=no ; disable of SMS reading from device when received
context = default ; context for incoming calls
group = 0 ; calling group
rxgain = 0 ; increase the incoming volume; may be negative
txgain = 0 ; increase the outgoint volume; may be negative
autodeletesms = yes ; auto delete incoming sms
resetquectel = yes ; reset quectel during initialization with ATZ command
u2diag = -1 ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres = yes ; use the caller ID presentation or not
callingpres = allowed_passed_screen ; set caller ID presentation by default use default network settings
disablesms = no ; disable of SMS reading from device when received
; chan_quectel has currently a bug with SMS reception. When a SMS gets in during a
; call chan_quectel might crash. Enable this option to disable sms reception.
; default = no
language=en_GB ; set channel default language
mindtmfgap=45 ; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=80 ; minimal DTMF tone duration in ms
mindtmfinterval=200 ; minimal interval between ends of DTMF of same digits in ms
language = en_GB ; set channel default language
mindtmfgap = 45 ; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration = 80 ; minimal DTMF tone duration in ms
mindtmfinterval = 200 ; minimal interval between ends of DTMF of same digits in ms
callwaiting=auto ; if 'yes' allow incoming calls waiting; by default use network settings
callwaiting = auto ; if 'yes' allow incoming calls waiting; by default use network settings
; if 'no' waiting calls just ignored
disable=no ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section
disable = no ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section
initstate=start ; specified initial state of device, must be one of 'stop' 'start' 'remote'
initstate = start ; specified initial state of device, must be one of 'stop' 'start' 'remote'
; 'remove' same as 'disable=yes'
exten=+1234567890 ; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)
exten = +1234567890 ; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)
dtmf=relax ; control of incoming DTMF detection, possible values:
dtmf = relax ; control of incoming DTMF detection, possible values:
; off - off DTMF tones detection, voice data passed to asterisk unaltered
; use this value for gateways or if not use DTMF for AVR or inside dialplan
; inband - do DTMF tones detection
@ -78,17 +78,8 @@ dtmf=relax ; control of incoming DTMF detection, possible values:
; quectel required settings
[quectel0]
context=quectel0
exten=+447123456789
context = quectel0
exten = +447123456789
audio=/dev/ttyUSB1 ; tty port for audio connection; no default value
data=/dev/ttyUSB2 ; tty port for AT commands; no default value
; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345
; imei and imsi must contain exactly 15 digits !
; imei/imsi discovery is available on Linux only
;imei=123456789012345
;imsi=123456789012345
; if audio and data set together with imei and/or imsi audio and data has precedence
; you can use both imei and imsi together in this case exact match by imei and imsi required
audio = /dev/ttyUSB1 ; tty port for audio connection; no default value
data = /dev/ttyUSB2 ; tty port for AT commands; no default value

4
asterisk/conf/mobile/res_stun_monitor.conf
View File

@ -5,5 +5,5 @@
;
[general]
stunaddr=stun.aa.net.uk:3478
stunrefresh=30
stunaddr = stun.aa.net.uk:3478
stunrefresh = 30

4
asterisk/conf/mobile/rtp.conf
View File

@ -4,5 +4,5 @@
; Inspired by/Stolen from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
;
[general]
rtpstart=62535
rtpend=65533
rtpstart = 62535
rtpend = 65533