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JerryXiao 2021-11-19 15:52:42 +08:00
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Signed by: Jerry
GPG key ID: 22618F758B5BE2E5
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233
codecs.conf Normal file
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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

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confbridge.conf Normal file
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[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

127
extensions.conf Normal file
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[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)

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modules.conf Normal file
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;
; 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]

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musiconhold.conf Normal file
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[default]
mode=files
directory=moh
;[custom]
;mode=files
;directory=/var/lib/asterisk/moh

144
pjsip.conf Normal file
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[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

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#!/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()