first commit
This commit is contained in:
commit
1e8fb5e9c0
GPG Key ID:
22618F758B5BE2E5
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@ -0,0 +1,233 @@
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[speex]
|
||||
; CBR encoding quality [0..10]
|
||||
; used only when vbr = false
|
||||
quality => 3
|
||||
|
||||
; codec complexity [0..10]
|
||||
; tradeoff between cpu/quality
|
||||
complexity => 2
|
||||
|
||||
; perceptual enhancement [true / false]
|
||||
; improves clarity of decoded speech
|
||||
enhancement => true
|
||||
|
||||
; voice activity detection [true / false]
|
||||
; reduces bitrate when no voice detected, used only for CBR
|
||||
; (implicit in VBR/ABR)
|
||||
vad => true
|
||||
|
||||
; variable bit rate [true / false]
|
||||
; uses bit rate proportionate to voice complexity
|
||||
vbr => true
|
||||
|
||||
; available bit rate [bps, 0 = off]
|
||||
; encoding quality modulated to match this target bit rate
|
||||
; not recommended with dtx or pp_vad - may cause bandwidth spikes
|
||||
abr => 0
|
||||
|
||||
; VBR encoding quality [0-10]
|
||||
; floating-point values allowed
|
||||
vbr_quality => 4
|
||||
|
||||
; discontinuous transmission [true / false]
|
||||
; stops transmitting completely when silence is detected
|
||||
; pp_vad is far more effective but more CPU intensive
|
||||
dtx => false
|
||||
|
||||
; preprocessor configuration
|
||||
; these options only affect Speex v1.1.8 or newer
|
||||
|
||||
; enable preprocessor [true / false]
|
||||
; allows dsp functionality below but incurs CPU overhead
|
||||
preprocess => false
|
||||
|
||||
; preproc voice activity detection [true / false]
|
||||
; more advanced equivalent of DTX, based on voice frequencies
|
||||
pp_vad => false
|
||||
|
||||
; preproc automatic gain control [true / false]
|
||||
pp_agc => false
|
||||
pp_agc_level => 8000
|
||||
|
||||
; preproc denoiser [true / false]
|
||||
pp_denoise => false
|
||||
|
||||
; preproc dereverb [true / false]
|
||||
pp_dereverb => false
|
||||
pp_dereverb_decay => 0.4
|
||||
pp_dereverb_level => 0.3
|
||||
|
||||
; experimental bitrate changes depending on RTCP feedback [true / false]
|
||||
experimental_rtcp_feedback => false
|
||||
|
||||
|
||||
[plc]
|
||||
; for all codecs which do not support native PLC
|
||||
; this determines whether to perform generic PLC
|
||||
; there is a minor performance penalty for this.
|
||||
; By default plc is applied only when the 2 codecs
|
||||
; in a channel are different.
|
||||
genericplc => true
|
||||
; Apply generic plc to channels even if the 2 codecs
|
||||
; are the same. This forces transcoding via slin so
|
||||
; the performance impact should be considered.
|
||||
; Ignored if genericplc is not also enabled.
|
||||
genericplc_on_equal_codecs => false
|
||||
|
||||
; Generate custom formats for formats requiring attributes.
|
||||
; After defining the custom format, the name used in defining
|
||||
; the format can be used throughout Asterisk in the format 'allow'
|
||||
; and 'disallow' options.
|
||||
;
|
||||
; Example: silk8 is a predefined custom format in this config file.
|
||||
; Once this config file is loaded, silk8 can be used anywhere a
|
||||
; peer's codec capabilities are defined.
|
||||
;
|
||||
; In sip.conf 'silk8' can be defined as a capability for a peer.
|
||||
; [peer1]
|
||||
; type=peer
|
||||
; host=dynamic
|
||||
; disallow=all
|
||||
; allow=silk8 ;custom codec defined in codecs.conf
|
||||
;
|
||||
; LIMITATIONS
|
||||
; Custom formats can only be defined at startup. Any changes to this
|
||||
; file made after startup will not take into effect until after Asterisk
|
||||
; is restarted.
|
||||
;
|
||||
|
||||
; Default Custom SILK format definitions, only one custom SILK format per
|
||||
; sample rate is allowed.
|
||||
[silk8]
|
||||
type=silk
|
||||
samprate=8000
|
||||
fec=true ; turn on or off encoding with forward error correction.
|
||||
; On recommended, off by default.
|
||||
packetloss_percentage=10 ; Estimated packet loss percentage in uplink direction. This
|
||||
; affects how much redundancy is built in when using fec.
|
||||
; The higher the percentage, the larger amount of bandwidth is
|
||||
; used. Default is 0%, 10% is recommended when fec is in use.
|
||||
|
||||
maxbitrate=10000 ; Use the table below to make sure a useful bitrate is choosen
|
||||
; for maxbitrate. If not set or value is not within the bounds
|
||||
; of the encoder, a default value is chosen.
|
||||
;
|
||||
; sample rate | bitrate range
|
||||
; 8khz | 5000 - 20000 bps
|
||||
; 12khz | 7000 - 25000 bps
|
||||
; 16khz | 8000 - 30000 bps
|
||||
; 24khz | 20000- 40000 bps
|
||||
;
|
||||
;dtx=true ; Encode using discontinuous transmission mode or not. Turning this
|
||||
; on will save bandwidth during periods of silence at the cost of
|
||||
; increased computational complexity. Off by default.
|
||||
|
||||
[silk12]
|
||||
type=silk
|
||||
samprate=12000
|
||||
maxbitrate=12000
|
||||
fec=true
|
||||
packetloss_percentage=10;
|
||||
|
||||
[silk16]
|
||||
type=silk
|
||||
samprate=16000
|
||||
maxbitrate=20000
|
||||
fec=true
|
||||
packetloss_percentage=10;
|
||||
|
||||
[silk24]
|
||||
type=silk
|
||||
samprate=24000
|
||||
maxbitrate=30000
|
||||
fec=true
|
||||
packetloss_percentage=10;
|
||||
|
||||
|
||||
; Default custom CELT codec definitions. Only one custom CELT definition is allowed
|
||||
; per a sample rate.
|
||||
;[celt44]
|
||||
;type=celt
|
||||
;samprate=44100 ; The samplerate in hz. This option is required.
|
||||
;framesize=480 ; The framesize option represents the duration of each frame in samples.
|
||||
; This must be a factor of 2. This option is only advertised in an SDP
|
||||
; when it is set. Otherwise a default of framesize of 480 is assumed
|
||||
; internally
|
||||
|
||||
;[celt48]
|
||||
;type=celt
|
||||
;samprate=48000
|
||||
|
||||
;[celt32]
|
||||
;type=celt
|
||||
;samprate=32000
|
||||
|
||||
;============================ OPUS Section Options ============================
|
||||
;
|
||||
; NOTE: Accurate documentation corresponding to your downloaded version of
|
||||
; codec_opus is available from Asterisk's CLI:
|
||||
;
|
||||
; *CLI> config show help codec_opus opus
|
||||
;
|
||||
;[opus]
|
||||
;type= ; Must be of type "opus" (default: "")
|
||||
;packet_loss= ; Encoder's packet loss percentage. Can be any number between 0
|
||||
; and 100, inclusive. A higher value results in more loss
|
||||
; resistance. (default: 0)
|
||||
;complexity= ; Encoder's computational complexity. Can be any number between 0
|
||||
; and 10, inclusive. Note, 10 equals the highest complexity.
|
||||
; (default: 10)
|
||||
;max_bandwidth= ; Encoder's maximum bandwidth allowed. Sets an upper bandwidth
|
||||
; bound on the encoder. Can be any of the following: narrow,
|
||||
; medium, wide, super_wide, full. (default: full)
|
||||
;signal= ; Encoder's signal type. Aids in mode selection on the encoder: Can
|
||||
; be any of the following: auto, voice, music. (default: auto)
|
||||
;application= ; Encoder's application type. Can be any of the following: voip,
|
||||
; audio, low_delay. (default: voip)
|
||||
;max_playback_rate= ; Override the maximum playback rate in the offer's SDP.
|
||||
; Any value between 8000 and 48000 (inclusive) is valid,
|
||||
; however typically it should match one of the usual opus
|
||||
; bandwidths. (default: 48000)
|
||||
;bitrate= ; Override the maximum average bitrate in the offer's SDP. Any value
|
||||
; between 500 and 512000 is valid. The following values are also
|
||||
; allowed: auto, max. (default: auto)
|
||||
;cbr= ; Override the constant bit rate parameter in the offer's SDP. A value of
|
||||
; 0/false/no represents a variable bit rate whereas 1/true/yes represents
|
||||
; a constant bit rate. (default: no)
|
||||
;fec= ; Override the use inband fec parameter in the offer's SDP. A value of
|
||||
; 0/false/no represents disabled whereas 1/true/yes represents enabled.
|
||||
; (default: yes)
|
||||
;dtx= ; Override the use dtx parameter in the offer's SDP. A value of 0/false/no
|
||||
; represents disabled whereas 1/true/yes represents enabled. (default: no)
|
||||
|
||||
;=============================== OPUS Examples ================================
|
||||
;
|
||||
;[opus]
|
||||
;type=opus
|
||||
;max_playback_rate=8000 ; Limit the maximum playback rate on the encoder
|
||||
;fec=no ; No inband fec
|
||||
|
||||
;[myopus]
|
||||
;type=opus
|
||||
;max_bandwidth=wide ; Maximum encoded bandwidth set to wide band (0-8000 Hz
|
||||
; ; audio bandwidth at 16Khz sample rate)
|
||||
;cbr=yes ; Negotiate a constant bit rate
|
||||
|
||||
[opus48]
|
||||
type=opus
|
||||
max_playback_rate=48000
|
||||
max_bandwidth=wide
|
||||
bitrate=48000
|
||||
packet_loss=10
|
||||
|
||||
[opus64]
|
||||
type=opus
|
||||
max_playback_rate=48000
|
||||
max_bandwidth=full
|
||||
bitrate=64000
|
||||
|
||||
[opus96]
|
||||
type=opus
|
||||
max_playback_rate=48000
|
||||
max_bandwidth=full
|
||||
bitrate=96000
|
||||
|
|
@ -0,0 +1,47 @@
|
|||
[sample_user_menu]
|
||||
type=menu
|
||||
*=playback_and_continue(conf-usermenu)
|
||||
*1=toggle_mute
|
||||
1=toggle_mute
|
||||
*4=decrease_listening_volume
|
||||
4=decrease_listening_volume
|
||||
*6=increase_listening_volume
|
||||
6=increase_listening_volume
|
||||
*7=decrease_talking_volume
|
||||
7=decrease_talking_volume
|
||||
*8=leave_conference
|
||||
8=leave_conference
|
||||
*9=increase_talking_volume
|
||||
9=increase_talking_volume
|
||||
|
||||
[sample_admin_menu]
|
||||
type=menu
|
||||
*=playback_and_continue(conf-adminmenu)
|
||||
*1=toggle_mute
|
||||
1=toggle_mute
|
||||
*2=admin_toggle_conference_lock ; only applied to admin users
|
||||
2=admin_toggle_conference_lock ; only applied to admin users
|
||||
*3=admin_kick_last ; only applied to admin users
|
||||
3=admin_kick_last ; only applied to admin users
|
||||
*4=decrease_listening_volume
|
||||
4=decrease_listening_volume
|
||||
*6=increase_listening_volume
|
||||
6=increase_listening_volume
|
||||
*7=decrease_talking_volume
|
||||
7=decrease_talking_volume
|
||||
*8=no_op
|
||||
8=no_op
|
||||
*9=increase_talking_volume
|
||||
9=increase_talking_volume
|
||||
|
||||
[user1]
|
||||
type=user
|
||||
;pin=1234
|
||||
marked=yes
|
||||
admin=no
|
||||
music_on_hold_when_empty=yes
|
||||
dsp_drop_silence=yes
|
||||
|
||||
[bridge1]
|
||||
type=bridge
|
||||
max_members=10
|
||||
|
|
@ -0,0 +1,127 @@
|
|||
[general]
|
||||
static=yes
|
||||
writeprotect=no
|
||||
clearglobalvars=no
|
||||
|
||||
[extdn42whois]
|
||||
;exten => i,1,NoOp()
|
||||
;exten => t,1,Goto(s,1)
|
||||
|
||||
exten => _X.,5,Set(CALLNUM=${EXTEN})
|
||||
same => n,Goto(s,1)
|
||||
|
||||
exten => s,1,Playback(silence/1)
|
||||
same => n,Set(USERINPUT=)
|
||||
same => n,Read(USERINPUT,jerry-whois,10,,1,6)
|
||||
same => n,GotoIf($["${READSTATUS}" = "TIMEOUT"]?s,1:)
|
||||
same => n,GotoIf($["${USERINPUT}" = ""]?extmymenu,${CALLNUM},5:whois,1)
|
||||
|
||||
exten => whois,1,Set(SESSIONUNID=${RAND(0,100000)})
|
||||
same => n,TrySystem(/var/lib/asterisk/scripts/jerry/whois.py ${USERINPUT} ${SESSIONUNID})
|
||||
same => n,Playback(/var/tmp/ast-dynamic/${SESSIONUNID})
|
||||
same => n,GotoIf($["${PLAYBACKSTATUS}" = "SUCCESS"]?whois,whoisend:)
|
||||
same => n,Playback(im-sorry&something-terribly-wrong)
|
||||
same => n(whoisend),Goto(s,1)
|
||||
|
||||
[extmymenu]
|
||||
exten => i,1,Playback(silence/1&goodbye)
|
||||
same => n,Hangup()
|
||||
|
||||
exten => _X.,5,Set(CALLNUM=${EXTEN})
|
||||
same => n,Goto(s,1)
|
||||
|
||||
exten => s,1,Wait(1)
|
||||
;same => n(loop),Background(vm-press&letters/a&number)
|
||||
same => n(loop),Background(jerry-intro)
|
||||
same => n,WaitExten(15)
|
||||
|
||||
exten => t,1,Goto(s,loop)
|
||||
|
||||
exten => _X,1,NoOp()
|
||||
;same => n,Playback(silence/1&you-entered)
|
||||
;same => n,SayNumber(${EXTEN})
|
||||
same => n,Wait(1)
|
||||
same => n,Goto(${EXTEN},100)
|
||||
|
||||
exten => 1,100,NoOp()
|
||||
same => n,Goto(extdn42whois,${CALLNUM},5)
|
||||
same => n,Goto(menuend,1)
|
||||
|
||||
exten => 2,100,NoOp()
|
||||
same => n,ConfBridge(1,bridge1,user1,sample_user_menu)
|
||||
same => n,Goto(menuend,1)
|
||||
|
||||
exten => 3,100,NoOp()
|
||||
same => n,Playback(your&number&is)
|
||||
same => n,SayAlpha(${CALLERID(num)})
|
||||
same => n,Playback(silence/1&calling)
|
||||
same => n,SayAlpha(${CALLNUM})
|
||||
same => n,Goto(menuend,1)
|
||||
|
||||
exten => 4,100,NoOp()
|
||||
same => n,Read(TMPNOM,z-external,1,,1,0.1)
|
||||
;same => n,Playback(z-external)
|
||||
same => n,Goto(menuend,1)
|
||||
|
||||
exten => 5,100,NoOp()
|
||||
same => n,Read(TMPNOM,z-macroform-cold_day,1,,1,0.1)
|
||||
;same => n,Playback(z-macroform-cold_day)
|
||||
same => n,Goto(menuend,1)
|
||||
|
||||
exten => _X,100,NoOp()
|
||||
same => n,Goto(menuend,1)
|
||||
|
||||
exten => menuend,1,NoOp()
|
||||
same => n,Wait(1)
|
||||
same => n,Goto(s,loop)
|
||||
|
||||
[extmyself]
|
||||
exten => i,1,NoOp()
|
||||
|
||||
exten => chanunavail,1,Playback(im-sorry&number-not-answering&please-try-call-later)
|
||||
exten => chanunavail,2,Hangup()
|
||||
|
||||
exten => 424036180001,5,Dial(PJSIP/REDACTED,300,m)
|
||||
exten => 424036180002,5,Dial(PJSIP/REDACTED,300,m)
|
||||
exten => 424036180003,5,Dial(PJSIP/REDACTED,300,m)
|
||||
exten => 424036180004,5,Dial(PJSIP/REDACTED,300,m)
|
||||
exten => 424036180005,5,Dial(PJSIP/REDACTED,300,m)
|
||||
exten => _X.,6,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chanunavail,1:)
|
||||
exten => 424036180000,5,Goto(extmymenu,${EXTEN},5)
|
||||
exten => 424036183618,5,Playback(silence/1&your&number&is)
|
||||
same => 6,SayAlpha(${CALLERID(num)})
|
||||
same => 7,Playback(silence/1)
|
||||
;same => 8,SayAlpha(${CALLERID(name)})
|
||||
;same => 9,Playback(silence/1)
|
||||
exten => 424036184242,5,Goto(extdn42whois,${EXTEN},5)
|
||||
exten => _42403618XXXX,5,Playback(im-sorry&check-number-dial-again)
|
||||
|
||||
[extpeers]
|
||||
exten => _42403618XXXX,5,Answer()
|
||||
same => n,Goto(extmyself,${EXTEN},5)
|
||||
|
||||
exten => _42401332XXXX,5,NoOp()
|
||||
same => n,Dial(PJSIP/${EXTEN}@nia)
|
||||
;same => n,Dial(PJSIP/${EXTEN:-4}@nia)
|
||||
|
||||
exten => _42403315XXXX,5,NoOp()
|
||||
same => n,Dial(PJSIP/${EXTEN}@zane)
|
||||
|
||||
exten => _42403088XXXX,5,NoOp()
|
||||
same => n,Dial(PJSIP/${EXTEN}@sunnet)
|
||||
|
||||
exten => _42401353XXXX,5,NoOp()
|
||||
same => n,Dial(PJSIP/${EXTEN}@hertz)
|
||||
|
||||
exten => _42400119XXXX,5,NoOp()
|
||||
same => n,Dial(PJSIP/${EXTEN}@jrb0001)
|
||||
|
||||
[jerry]
|
||||
;exten => _X.,1,Set(CHANNEL(musicclass)=custom)
|
||||
exten => _X.,1,NoOp()
|
||||
|
||||
exten => _XXXX,2,Goto(42403618${EXTEN},1)
|
||||
exten => _XXXXXXXX,2,Goto(4240${EXTEN},1)
|
||||
exten => _X.,2,NoOp()
|
||||
|
||||
exten => _X.,3,Goto(extpeers,${EXTEN},5)
|
||||
|
|
@ -0,0 +1,83 @@
|
|||
;
|
||||
; Asterisk configuration file
|
||||
;
|
||||
; Module Loader configuration file
|
||||
;
|
||||
|
||||
[modules]
|
||||
autoload=yes
|
||||
;
|
||||
; Any modules that need to be loaded before the Asterisk core has been
|
||||
; initialized (just after the logger has been initialized) can be loaded
|
||||
; using 'preload'. This will frequently be needed if you wish to map all
|
||||
; module configuration files into Realtime storage, since the Realtime
|
||||
; driver will need to be loaded before the modules using those configuration
|
||||
; files are initialized.
|
||||
;
|
||||
; An example of loading ODBC support would be:
|
||||
;preload => res_odbc.so
|
||||
;preload => res_config_odbc.so
|
||||
;
|
||||
; If you want, load the GTK console right away.
|
||||
; Don't load the KDE console since
|
||||
; it's not as sophisticated right now.
|
||||
;
|
||||
noload => pbx_gtkconsole.so
|
||||
;load => pbx_gtkconsole.so
|
||||
noload => pbx_kdeconsole.so
|
||||
;
|
||||
; Intercom application is obsoleted by
|
||||
; chan_oss. Don't load it.
|
||||
;
|
||||
noload => app_intercom.so
|
||||
;
|
||||
; The 'modem' channel driver and its subdrivers are
|
||||
; obsolete, don't load them.
|
||||
;
|
||||
noload => chan_modem.so
|
||||
noload => chan_modem_aopen.so
|
||||
noload => chan_modem_bestdata.so
|
||||
noload => chan_modem_i4l.so
|
||||
;
|
||||
; Comment this out (after installing CAPI middleware and hardware
|
||||
; drivers) if you have CAPI-able hardware and wish to use it in
|
||||
; Asterisk.
|
||||
;
|
||||
noload => chan_capi.so
|
||||
;
|
||||
load => res_musiconhold.so
|
||||
;
|
||||
; Do not load load local channel drivers (using the system speaker) by default,
|
||||
; they are not used in most installations and might block the sound hardware
|
||||
;
|
||||
noload => chan_alsa.so
|
||||
noload => chan_console.so
|
||||
noload => chan_oss.so
|
||||
;
|
||||
; Disable CDR logging to SQLite by default since it writes unconditionally to
|
||||
; cdr.db without a way to rotate it.
|
||||
;
|
||||
noload => cdr_sqlite.so
|
||||
;
|
||||
; These conflict with app_directory.so and each other.
|
||||
noload => app_directory_odbc.so
|
||||
;
|
||||
; Enable these if you want to configure Asterisk in a database
|
||||
;
|
||||
noload => res_config_odbc.so
|
||||
noload => res_config_pgsql.so
|
||||
|
||||
noload => chan_sip.so
|
||||
;preload => codec_opus.so
|
||||
;preload => format_ogg_opus.so
|
||||
;preload => res_sorcery_config.so
|
||||
;preload => res_format_attr_opus.so
|
||||
;preload => codec_gsm.so
|
||||
;preload => format_gsm.so
|
||||
;preload => format_wav_gsm.so
|
||||
;preload => codec_resample.so
|
||||
;
|
||||
; Module names listed in "global" section will have symbols globally
|
||||
; exported to modules loaded after them.
|
||||
;
|
||||
[global]
|
||||
|
|
@ -0,0 +1,7 @@
|
|||
[default]
|
||||
mode=files
|
||||
directory=moh
|
||||
|
||||
;[custom]
|
||||
;mode=files
|
||||
;directory=/var/lib/asterisk/moh
|
||||
|
|
@ -0,0 +1,144 @@
|
|||
[global]
|
||||
type=global
|
||||
user_agent=PBX
|
||||
|
||||
[v4trans]
|
||||
type=transport
|
||||
protocol=udp
|
||||
bind=0.0.0.0:5060
|
||||
|
||||
[v6trans]
|
||||
type=transport
|
||||
protocol=udp
|
||||
bind=[::]:5060
|
||||
|
||||
;=============TEMPLATE==============
|
||||
|
||||
[endpoint-template](!)
|
||||
type=endpoint
|
||||
context=jerry
|
||||
;message_context=jerry-msg
|
||||
allow=!all,ulaw,opus48,opus,speex,alaw
|
||||
;direct_media=no
|
||||
rtp_symmetric=no
|
||||
force_rport=yes
|
||||
rewrite_contact=yes
|
||||
|
||||
[auth-template](!)
|
||||
type=auth
|
||||
|
||||
[aor-template](!)
|
||||
type=aor
|
||||
max_contacts=1
|
||||
remove_existing=yes
|
||||
|
||||
;=============PHONES==============
|
||||
|
||||
[REDACTED](endpoint-template)
|
||||
auth = REDACTED
|
||||
aors = REDACTED
|
||||
allow=!all,opus48,opus,ulaw,speex,alaw
|
||||
callerid=Jerry <424036180001>
|
||||
[REDACTED](auth-template)
|
||||
auth_type=userpass
|
||||
username=REDACTED
|
||||
password=REDACTED
|
||||
[REDACTED](aor-template)
|
||||
|
||||
[REDACTED](endpoint-template)
|
||||
auth = REDACTED
|
||||
aors = REDACTED
|
||||
allow=!all,opus48,opus,ulaw,speex,alaw
|
||||
callerid=Jerry <424036180002>
|
||||
[REDACTED](auth-template)
|
||||
auth_type=userpass
|
||||
username=REDACTED
|
||||
password=REDACTED
|
||||
[REDACTED](aor-template)
|
||||
|
||||
[REDACTED](endpoint-template)
|
||||
auth = REDACTED
|
||||
aors = REDACTED
|
||||
allow=!all,opus48,opus,ulaw,speex,alaw
|
||||
callerid=Jerry <424036180003>
|
||||
[REDACTED](auth-template)
|
||||
auth_type=userpass
|
||||
username=REDACTED
|
||||
password=REDACTED
|
||||
[REDACTED](aor-template)
|
||||
|
||||
[REDACTED](endpoint-template)
|
||||
auth = REDACTED
|
||||
aors = REDACTED
|
||||
allow=!all,opus48,opus,ulaw,speex,alaw
|
||||
callerid=Jerry <424036180004>
|
||||
[REDACTED](auth-template)
|
||||
auth_type=userpass
|
||||
username=REDACTED
|
||||
password=REDACTED
|
||||
[REDACTED](aor-template)
|
||||
|
||||
[REDACTED](endpoint-template)
|
||||
auth = REDACTED
|
||||
aors = REDACTED
|
||||
allow=!all,opus48,opus,ulaw,speex,alaw
|
||||
callerid=Jerry <424036180005>
|
||||
[REDACTED](auth-template)
|
||||
auth_type=userpass
|
||||
username=REDACTED
|
||||
password=REDACTED
|
||||
[REDACTED](aor-template)
|
||||
|
||||
;=============EXTERN==============
|
||||
|
||||
;=============PEERS==============
|
||||
|
||||
[nia](endpoint-template)
|
||||
aors = nia
|
||||
identify_by=ip
|
||||
[nia](aor-template)
|
||||
contact=sip:172.20.168.194:5160
|
||||
[nia]
|
||||
type=identify
|
||||
endpoint=nia
|
||||
match=172.20.168.194
|
||||
|
||||
[zane](endpoint-template)
|
||||
aors = zane
|
||||
identify_by=ip
|
||||
[zane](aor-template)
|
||||
contact=sip:172.22.167.72:5060
|
||||
[zane]
|
||||
type=identify
|
||||
endpoint=zane
|
||||
match=172.22.167.72
|
||||
|
||||
[sunnet](endpoint-template)
|
||||
aors = sunnet
|
||||
identify_by=ip
|
||||
[sunnet](aor-template)
|
||||
contact=sip:10.127.11.130:5060
|
||||
[sunnet]
|
||||
type=identify
|
||||
endpoint=sunnet
|
||||
match=10.127.11.130
|
||||
|
||||
[hertz](endpoint-template)
|
||||
aors = hertz
|
||||
identify_by=ip
|
||||
[hertz](aor-template)
|
||||
contact=sip:172.20.29.73:5060
|
||||
[hertz]
|
||||
type=identify
|
||||
endpoint=hertz
|
||||
match=172.20.29.73
|
||||
|
||||
[jrb0001](endpoint-template)
|
||||
aors = jrb0001
|
||||
identify_by=ip
|
||||
[jrb0001](aor-template)
|
||||
contact=sip:[fd42:5d71:219:1003:216:3eff:fe9d:882f]:5060
|
||||
[jrb0001]
|
||||
type=identify
|
||||
endpoint=jrb0001
|
||||
match=fd42:5d71:219:1003:216:3eff:fe9d:882f
|
||||
|
|
@ -0,0 +1,137 @@
|
|||
#!/usr/bin/python
|
||||
|
||||
import socket
|
||||
import subprocess
|
||||
from pathlib import Path
|
||||
from sys import argv
|
||||
import re
|
||||
import shutil
|
||||
from time import time
|
||||
|
||||
WHOIS_SERVER = ('172.20.129.8', 43)
|
||||
ASTROOT = Path("/var/lib/asterisk/sounds")
|
||||
ERRSOUND = ASTROOT / "en" / "something-terribly-wrong.gsm"
|
||||
ROOTDIR = Path("/var/tmp")
|
||||
OUTDIR = ROOTDIR / "ast-dynamic"
|
||||
OUTDIR.mkdir(exist_ok=True)
|
||||
|
||||
def cleanup():
|
||||
try:
|
||||
now = time()
|
||||
for f in OUTDIR.iterdir():
|
||||
if f.is_file() and now - f.stat().st_mtime > 60.0:
|
||||
f.unlink()
|
||||
except Exception:
|
||||
raise
|
||||
|
||||
def espeak(to_speak, unid):
|
||||
p1 = subprocess.run(
|
||||
["espeak-ng", "-v", "en-us", "--stdin", "-p", "80", "--stdout"],
|
||||
input=str(to_speak).encode("utf-8"),
|
||||
capture_output=True,
|
||||
check=True,
|
||||
timeout=10
|
||||
)
|
||||
unfile = f"{unid}.wav"
|
||||
p2 = subprocess.run(['ffmpeg', '-hide_banner', '-i', "-",
|
||||
'-vn', "-ac", "1", "-ar", "8000", "-y",
|
||||
f"{OUTDIR / unfile}"],
|
||||
input=p1.stdout,
|
||||
capture_output=True,
|
||||
check=True,
|
||||
timeout=10
|
||||
)
|
||||
|
||||
def whois(arg):
|
||||
try:
|
||||
i = int(arg)
|
||||
except ValueError:
|
||||
raise ValueError(f"invalid autonomous system number {','.join(list(arg))}.")
|
||||
else:
|
||||
if 0 <= i <= 9999:
|
||||
arg = f"AS424242{i:04d}"
|
||||
elif 20000 <= i <= 29999:
|
||||
arg = f"AS42424{i:05d}"
|
||||
elif 70000 <= i <= 79999:
|
||||
arg = f"AS42012{i:05d}"
|
||||
else:
|
||||
arg = f"AS{i}"
|
||||
with socket.socket(family=socket.AF_INET, type=socket.SOCK_STREAM) as s:
|
||||
s.settimeout(5)
|
||||
s.connect(WHOIS_SERVER)
|
||||
s.send((f"{arg}\r\n").encode("utf-8"))
|
||||
r = bytes()
|
||||
while len(_r := s.recv(1024)) != 0:
|
||||
r += _r
|
||||
r = r.decode('utf-8')
|
||||
lines = r.split('\n')
|
||||
target = None
|
||||
for (i, line) in enumerate(lines[::-1]):
|
||||
if line:
|
||||
target = False
|
||||
elif not line and target is False:
|
||||
target = -i
|
||||
break
|
||||
lines = lines[target:]
|
||||
def multiline_process():
|
||||
attr = ""
|
||||
for i, line in enumerate(lines):
|
||||
if not line:
|
||||
continue
|
||||
if line.startswith('%'):
|
||||
continue
|
||||
if m := re.match(r'^([\S]+):.*$', line):
|
||||
attr = m.groups()[0]
|
||||
assert attr
|
||||
else:
|
||||
assert attr
|
||||
lines[i] = f"{attr}:{lines[i]}"
|
||||
try:
|
||||
multiline_process()
|
||||
except Exception as e:
|
||||
raise
|
||||
whois_filter = lambda x: any([x.startswith(field) for field in {'remarks', 'nserver', 'descr', 'ds-rdata', 'mp-import', 'mp-export'}])
|
||||
lines = [l for l in lines if l and not whois_filter(l)]
|
||||
lines_new = list()
|
||||
repl = {
|
||||
"aut-num": "Autonomous System Number",
|
||||
"as-name": "Autonomous System Name",
|
||||
"admin-c": "Administrative Contact",
|
||||
"tech-c": "Technical Contact",
|
||||
"mnt-by": "Maintained by",
|
||||
}
|
||||
for l in lines:
|
||||
if not l.strip():
|
||||
continue
|
||||
elif l.startswith('%'):
|
||||
continue
|
||||
elif l.startswith("aut-num:"):
|
||||
_p, _s = l.split()
|
||||
_s = ",".join(list(_s)).replace("A,S,", "")
|
||||
l = f"{_p} {_s}"
|
||||
for fr, to in repl.items():
|
||||
if l.startswith(f"{fr}:"):
|
||||
l = l.replace(f"{fr}:", f"{to}:", 1)
|
||||
lines_new.append(l)
|
||||
|
||||
if not lines_new:
|
||||
lines_new.append(f"There is no such Autonomous System {','.join(list(arg)).replace('A,S,', '')}")
|
||||
else:
|
||||
lines_new.insert(0, "Found the following who is record from the dn42 registry.")
|
||||
return ".\n".join(lines_new)+"."
|
||||
|
||||
def main():
|
||||
# argv[1]: as number
|
||||
# argv[2]: unique identifier
|
||||
cleanup()
|
||||
try:
|
||||
ret = whois(argv[1])
|
||||
except Exception as e:
|
||||
ret = f"error, {e}"
|
||||
try:
|
||||
espeak(ret, argv[2])
|
||||
except Exception:
|
||||
outpath = OUTDIR / f"{argv[2]}{ERRSOUND.suffix}"
|
||||
shutil.copy(ERRSOUND, outpath)
|
||||
if __name__ == "__main__":
|
||||
main()
|
||||
Loading…
Reference in New Issue