asterisk/asterisk.txt@a2c6460cfb16
asterisk/asterisk.txt
Mon, 28 Jan 2013 17:37:18 +0100
- author
- Michael Schloh von Bennewitz <michael@schloh.com>
- date
- Mon, 28 Jan 2013 17:37:18 +0100
- changeset 758
- a2c6460cfb16
- parent 399
- 9360a38b5c9f
- permissions
- -rw-r--r--
Correct socket error reporting improvement with IPv6 portable code,
after helpful recommendation by Saúl Ibarra Corretgé on OSips devlist.
1 <file name="asterisk.conf">
2 ;;
3 ;; asterisk.conf -- Asterisk master configuration
4 ;;
6 [directories]
7 astetcdir = @l_prefix@/etc/asterisk
8 astmoddir = @l_prefix@/lib/asterisk/modules
9 astagidir = @l_prefix@/lib/asterisk/agi-bin
10 astvarlibdir = @l_prefix@/share/asterisk
11 astspooldir = @l_prefix@/var/asterisk/spool
12 astrundir = @l_prefix@/var/asterisk/run
13 astlogdir = @l_prefix@/var/asterisk/log
14 astdbdir = @l_prefix@/var/asterisk/db
16 [files]
17 astctlowner = @l_rusr@
18 astctlgroup = @l_rgrp@
19 astctlpermissions = 700
20 astctl = asterisk.ctl
22 [options]
23 systemname = openpkg-pbx
24 runuser = @l_rusr@
25 rungroup = @l_rgrp@
26 verbose = 0
27 alwaysfork = yes
28 dumpcore = no
29 quiet = yes
30 highpriority = no
31 initcrypto = no
32 nocolor = yes
33 execincludes = no
34 ;timestamp = yes
35 ;optiondebug = no
36 ;nofork = no
37 ;console = no
38 ;dontwarn = no
40 </file>
41 <file name="amd.conf">
42 ;;
43 ;; amd.conf -- Answering Machine Detection configuration
44 ;;
46 [general]
47 initial_silence = 2500 ; Maximum silence duration before the greeting.
48 ; If exceeded then MACHINE.
49 greeting = 1500 ; Maximum length of a greeting.
50 ;If exceeded then MACHINE.
51 after_greeting_silence = 800 ; Silence after detecting a greeting.
52 ; If exceeded then HUMAN.
53 total_analysis_time = 5000 ; Maximum time allowed for the algorithm to
54 ; decide on a HUMAN or MACHINE.
55 min_word_length = 100 ; Minimum duration of Voice to considered a word.
56 between_words_silence = 50 ; Minimum duration of silence after a word to
57 ; consider the audio what follows as a new word.
58 maximum_number_of_words = 3 ; Maximum number of words in the greeting.
59 ; If exceeded then MACHINE.
60 silence_threshold = 256
62 </file>
63 <file name="modules.conf">
64 ;;
65 ;; modules.conf -- Asterisk functionality module configuration
66 ;;
68 [modules]
69 autoload = yes
71 [global]
73 </file>
74 <file name="logger.conf">
75 ;;
76 ;; logger.conf -- Asterisk logging configuration
77 ;;
79 [general]
80 dateformat = %F %T
81 queue_log = no
82 event_log = no
84 [logfiles]
85 console = error,warning,notice,verbose
86 asterisk.log = error,warning,notice ; verbose,debug
88 </file>
89 <file name="manager.conf">
90 ;;
91 ;; manager.conf -- Asterisk internal manager API configuration
92 ;;
94 [general]
95 enabled = yes
96 webenabled = no
97 bindaddr = 127.0.0.1
98 port = 5038
100 ; You can open a TLS connection to this socket with:
101 ;
102 ; openssl s_client -connect my_host:5039
103 ;
104 tlsenable = no
105 tlsbindaddr = 127.0.0.1
106 tlsbindport = 5039
107 tlscertfile = @l_prefix@/etc/asterisk/asterisk.pem
108 ; if tlsprivatekey is not specified search tlscertfile for key
109 ;tlsprivatekey = @l_prefix@/etc/asterisk/asterkey.pem
110 ;tlscipher = ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL
112 displayconnects = yes
113 allowmultiplelogin = yes
114 timestampevents = yes
116 [asterisk]
117 secret = asterisk
118 deny = 0.0.0.0/0.0.0.0
119 permit = 127.0.0.1/255.0.0.0
120 read = system,call,agent,user,config,log,verbose,dtmf,reporting,cdr,dialplan
121 write = system,call,agent,user,config,command,reporting
123 </file>
124 <file name="http.conf">
125 ;;
126 ;; http.conf -- Asterisk HTTP server interface
127 ;;
129 [general]
130 enabled = no
131 bindaddr = 127.0.0.1
132 bindport = 8088
133 tlsenable = no
134 tlsbindport = 8089
135 tlsbindaddr = 127.0.0.1
136 tlscertfile = @l_prefix@/etc/asterisk/asterisk.pem
137 enablestatic = yes
138 prefix = asterisk
139 redirect = / /asterisk/static/docs/index.html
141 [post_mappings]
142 uploads = @l_prefix@/var/asterisk/spool/uploads/
144 </file>
145 <file name="sip.conf">
146 ;;
147 ;; sip.conf -- Asterisk SIP configuration
148 ;;
150 [general]
151 useragent = OpenPKG Asterisk PBX
152 realm = example
153 bindaddr = 127.0.0.1
154 bindport = 5060
155 tcpenable = yes
156 tcpbindaddr = 127.0.0.1:5060
157 tlsenable = no
158 tlsbindaddr = 127.0.0.1:5061
159 tlscipher = ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL
160 tlscertfile = asterisk.pem
161 tlscafile = asterisk.pem
162 srvlookup = yes
163 useclientcode = yes
164 allowguest = yes
165 canreinvite = no
166 nat = no
167 disallow = all
168 allow = speex
169 allow = g726
170 allow = ulaw
171 allow = alaw
172 allow = gsm
173 videosupport = no
174 ;allow = h263
175 ;allow = h263p
176 notifyhold = yes
177 notifyringing = yes
178 limitonpeer = yes
179 call-limit = 1
180 incominglimit = 1
181 context = external
182 ;register = NNNNNNN:XXXXXX:NNNNNNN@sipgate.de/s
183 ;tos = 0x18
185 ;[sipgate]
186 ;type = peer
187 ;defaultuser = NNNNNNN
188 ;host = sipgate.de
189 ;fromuser = NNNNNNN
190 ;fromdomain = sipgate.de
191 ;canreinvite = no
192 ;disallow = all
193 ;allow = speex
194 ;allow = g726
195 ;allow = ulaw
196 ;allow = alaw
197 ;allow = gsm
198 ;context = external
200 ;[gw]
201 ;type = friend
202 ;defaultuser = gw
203 ;callerid = "ISDN-to-SIP" <gw>
204 ;fromdomain = example.com
205 ;secret = asterisk
206 ;host = dynamic
207 ;canreinvite = no
208 ;disallow = all
209 ;allow = g726
210 ;allow = ulaw
211 ;allow = alaw
212 ;allow = gsm
213 ;dtmfmode = rfc2833
214 ;qualify = yes
215 ;insecure = yes
216 ;context = external
217 ;nat = no
219 [std-user](!)
220 type = friend
221 context = internal
222 host = dynamic
223 dtmfmode = rfc2833
224 qualify = yes
225 disallow = all
226 allow = speex
227 allow = g726
228 allow = ulaw
229 allow = alaw
231 [behind-nat](!)
232 nat = yes
234 [with-mailbox](!)
235 hasvoicemail = yes
236 subscribemwi = yes
237 subscribecontext = internal
238 vmexten = voicemail
240 [foo](std-user,with-mailbox)
241 secret = asterisk
242 callerid = "Mr. Foo" <11>
243 mailbox = 11@internal
245 [bar](std-user,with-mailbox)
246 secret = asterisk
247 callerid = "Mr. Bar" <12>
248 mailbox = 12@internal
250 </file>
251 <file name="iax.conf">
252 ;;
253 ;; iax.conf -- Asterisk IAX configuration
254 ;;
256 ;; This configuration is reread at reload
257 ;; or with the CLI command
258 ;; reload chan_iax2.so
259 ;;
260 ;; General settings, like port number to bind to, and
261 ;; an option address (the default is to bind to all
262 ;; local addresses).
263 ;;
264 ;[general]
265 ;bindport=4569 ; bindport and bindaddr may be specified
266 ; ; NOTE: bindport must be specified BEFORE
267 ; ; bindaddr or may be specified on a specific
268 ; ; bindaddr if followed by colon and port
269 ; ; (e.g. bindaddr=192.168.0.1:4569)
270 ;bindaddr=127.0.0.1 ; more than once to bind to multiple
271 ; ; addresses, but the first will be the
272 ; ; default
274 </file>
275 <file name="iaxprov.conf">
276 ;;
277 ;; iaxprov.conf -- IAX2 provisioning information
278 ;;
280 ; Contains provisioning information for templates and for specific service
281 ; entries.
282 ;
283 ; Templates provide a group of settings from which provisioning takes place.
284 ; A template may be based upon any template that has been specified before
285 ; it. If the template that an entry is based on is not specified then it is
286 ; presumed to be 'default' (unless it is the first of course).
287 ;
288 ; Templates which begin with 'si-' are used for provisioning units with
289 ; specific service identifiers. For example the entry "si-000364000126"
290 ; would be used when the device with the corresponding service identifier of
291 ; "000364000126" attempts to register or make a call.
292 ;
293 [default]
294 port=4569 ; the port number the device should bind to (default 4569)
295 server=127.0.0.1 ; our PRIMARY server for registration and placing calls
297 ; altserver is the BACKUP server for registration and placing calls in the
298 ; event the primary server is unavailable.
299 ;
300 altserver=127.0.0.2
302 ; port is the port number to use for IAX2 outbound. The connections to the
303 ; server and altserver (default 4569)
304 ;
305 serverport=4569
306 language=es ; the preferred language for the device
307 codec=ulaw ; requested codec, the iaxy supports ulaw and adpcm
309 ; flags is a comma separated list of flags which the device should
310 ; use and may contain any of the following keywords:
311 ;
312 ; "register" - Register with server
313 ; "secure" - Do not accept calls / provisioning not originated by server
314 ; "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
315 ; "debug" - Output extra debugging to port 9999
316 ;
317 ; Note that use can use += and -= to adjust parameters
318 ;
319 flags=register
321 tos=ef ; see doc/ip-tos.txt
323 ; Example iaxy provisioning
324 ;
325 ;[si-000364000126]
326 ;user=iaxy
327 ;pass=bitsy
328 ;flags += debug
330 ;[si-000364000127]
331 ;user=iaxy2
332 ;pass=bitsy2
333 ;template=si-000364000126
334 ;flags += debug
337 ; If specified, the '*' provisioning is used for all devices which do not
338 ; have another provisioning entry within the file. If unspecified, no
339 ; provisioning will take place for devices which have no entry. DO NOT
340 ; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
341 ;
342 ;[*]
344 ;template=default
345 </file>
346 <file name="rtp.conf">
347 ;;
348 ;; rtp.conf -- Asterisk RTP configuration
349 ;;
351 [general]
352 rtpstart = 7070
353 rtpend = 7089
355 </file>
356 <file name="sip_notify.conf">
357 ;;
358 ;; sip_notify.conf -- Asterisk NOTIFY automation from command line
359 ;;
361 ; rfc3842
362 ; put empty "Content=>" at the end to have CRLF after last body line
363 [clear-mwi]
364 Event=>message-summary
365 Content-type=>application/simple-message-summary
366 Content=>Messages-Waiting: no
367 Content=>Message-Account: sip:asterisk@127.0.0.1
368 Content=>Voice-Message: 0/0 (0/0)
369 Content=>
371 ; Aastra
372 [aastra-check-cfg]
373 Event=>check-sync
375 [aastra-xml]
376 Event=>aastra-xml
378 ; Linksys
379 [linksys-cold-restart]
380 Event=>reboot_now
382 [linksys-warm-restart]
383 Event=>restart_now
385 ; Polycom
386 [polycom-check-cfg]
387 Event=>check-sync
389 ; Sipura
390 [sipura-check-cfg]
391 Event=>resync
393 [sipura-get-report]
394 Event=>report
396 ; Snom
397 [snom-check-cfg]
398 Event=>check-sync\;reboot=false
400 [snom-reboot]
401 Event=>reboot
403 ; Cisco
404 [cisco-check-cfg]
405 Event=>check-sync
407 </file>
408 <file name="extconfig.conf">
409 ;;
410 ;; extconfig.conf -- Static and realtime external configuration engine
411 ;;
413 [settings]
414 ;
415 ; Static configuration files:
416 ;
417 ; file.conf => driver,database[,table[,priority]]
418 ;
419 ; maps a particular configuration file to the given
420 ; database driver, database and table (or uses the
421 ; name of the file as the table if not specified)
422 ;
423 ;uncomment to load queues.conf via the odbc engine.
424 ;
425 ;queues.conf => odbc,asterisk,ast_config
426 ;extensions.conf => sqlite,asterisk,ast_config
427 ;
428 ; The following files CANNOT be loaded from Realtime storage:
429 ; asterisk.conf
430 ; extconfig.conf (this file)
431 ; logger.conf
432 ;
433 ; Additionally, the following files cannot be loaded from
434 ; Realtime storage unless the storage driver is loaded
435 ; early using 'preload' statements in modules.conf:
436 ; manager.conf
437 ; cdr.conf
438 ; rtp.conf
439 ;
440 ;
441 ; Realtime configuration engine
442 ;
443 ; maps a particular family of realtime
444 ; configuration to a given database driver,
445 ; database and table (or uses the name of
446 ; the family if the table is not specified
447 ;
448 ;example => odbc,asterisk,alttable,1
449 ;example => mysql,asterisk,alttable,2
450 ;example2 => ldap,"dc=oxymium,dc=net",example2
451 ;
452 ; Additionally, priorities are now supported for use as failover methods
453 ; for retrieving realtime data. If one connection fails to retrieve any
454 ; information, the next sequential priority will be tried next. This
455 ; especially works well with ODBC connections, since res_odbc now caches
456 ; when connection failures occur and prevents immediately retrying those
457 ; connections until after a specified timeout. Note: priorities must
458 ; start at 1 and be sequential (i.e. if you have only priorities 1, 2,
459 ; and 4, then 4 will be ignored, because there is no 3).
460 ;
461 ; "odbc" is shown in the examples below, but is not the only valid realtime
462 ; engine. There is:
463 ; odbc ... res_config_odbc
464 ; sqlite ... res_config_sqlite
465 ; pgsql ... res_config_pgsql
466 ; curl ... res_config_curl
467 ; ldap ... res_config_ldap
468 ;
469 ;iaxusers => odbc,asterisk
470 ;iaxpeers => odbc,asterisk
471 ;sippeers => odbc,asterisk
472 ;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
473 ;voicemail => odbc,asterisk
474 ;extensions => odbc,asterisk
475 ;meetme => mysql,general
476 ;queues => odbc,asterisk
477 ;queue_members => odbc,asterisk
478 ;musiconhold => mysql,general
479 ;queue_log => mysql,general
480 ;
481 ;
482 ; While most dynamic realtime engines are automatically used when defined in
483 ; this file, 'extensions', distinctively, is not. To activate dynamic realtime
484 ; extensions, you must turn them on in each respective context within
485 ; extensions.conf with a switch statement. The syntax is:
486 ; switch => Realtime/[[db_context@]tablename]/<opts>
487 ; The only option available currently is the 'p' option, which disallows
488 ; extension pattern queries to the database. If you have no patterns defined
489 ; in a particular context, this will save quite a bit of CPU time. However,
490 ; note that using dynamic realtime extensions is not recommended anymore as a
491 ; best practice; instead, you should consider writing a static dialplan with
492 ; proper data abstraction via a tool like func_odbc.
494 </file>
495 <file name="extensions.conf">
496 ;;
497 ;; extensions.conf -- Asterisk inbound & outbound call configuration
498 ;;
500 [general]
501 static = yes
502 writeprotect = yes
503 autofallthrough = yes
505 [globals]
506 MEETME_SPOOLDIR = @l_prefix@/var/asterisk/spool/meetme
507 STAFF = SIP/foo&SIP/bar
508 CONSOLE = Console/dsp
509 DOLLAR = $
511 ;;
512 ;; SPECIAL CONTEXTS
513 ;;
515 [macro-dial]
516 exten = s,1,Dial(${ARG1},${ARG2},${ARG3})
517 exten = s,n,Goto(s-${DIALSTATUS},1)
518 exten = s-ANSWER,1,Hangup
519 exten = s-BUSY,1,GotoIf($["${ARG4}" == ""]?novm)
520 exten = s-BUSY,n,GotoIf($[${MAILBOX_EXISTS(${ARG4})} == 0]?novm)
521 exten = s-BUSY,n,VoiceMail(${ARG4},b)
522 exten = s-BUSY,n,Playback(vm-goodbye)
523 exten = s-BUSY,n(novm),Hangup
524 exten = s-NOANSWER,1,GotoIf($["${ARG4}" == ""]?novm)
525 exten = s-NOANSWER,n,MailboxExists(${ARG4})
526 exten = s-NOANSWER,n,GotoIf($[${MAILBOX_EXISTS(${ARG4})} == 0]?novm)
527 exten = s-NOANSWER,n,VoiceMail(${ARG4},u)
528 exten = s-NOANSWER,n,Playback(vm-goodbye)
529 exten = s-NOANSWER,n(novm),Hangup
530 exten = _s-.,1,Goto(s-NOANSWER,1)
532 [default]
533 ; currently empty
535 ;;
536 ;; EXTERNAL DIAL CONTEXT
537 ;;
539 [external]
540 include = default
542 ; external incoming SIP connection
543 exten = example,hint,${STAFF}
544 exten = example,1,Goto(s,1)
545 exten = s,n,Ringing
546 exten = s,n,Wait(1)
547 exten = s,n,Answer
548 exten = s,n,Macro(dial,${STAFF},30,gTtr,1@external)
550 ; external to internal mapping
551 exten = foo,hint,SIP/foo
552 exten = foo,1,Goto(internal,foo,1)
553 exten = bar,hint,SIP/bar
554 exten = bar,1,Goto(internal,bar,1)
556 ;;
557 ;; INTERNAL DIAL CONTEXT
558 ;;
560 [internal]
561 include = default
562 ;include = parkedcalls
564 ; internal to external mapping
565 exten = example,1,Goto(external,example,1)
567 ; internal user <foo> #11
568 exten = foo,hint,SIP/foo
569 exten = foo,1,Goto(11,1)
570 exten = 11,hint,SIP/foo
571 exten = 11,1,Macro(dial,SIP/foo,30,gTtr,11@internal)
573 ; internal user <bar> #12
574 exten = bar,hint,SIP/bar
575 exten = bar,1,Goto(12,1)
576 exten = 12,hint,SIP/bar
577 exten = 12,1,Macro(dial,SIP/bar,30,gTtr,12@internal)
579 ; internal group <all> #20
580 exten = all,1,Goto(20,1)
581 exten = 20/foo,1,Macro(dial,SIP/bar,60,)
582 exten = 20/bar,1,Macro(dial,SIP/foo,60,)
584 ; internal service <conference> #7<n>
585 exten = conference,1,Goto(70,1)
586 exten = _7[0-9],1,Set(confno=${EXTEN:1})
587 exten = _7[0-9],n,Goto(7,enter)
588 exten = 7,1,Set(TIMEOUT(digit)=3)
589 exten = 7,n,Set(TIMEOUT(response)=6)
590 exten = 7,n(repeat),Read(confno,conf-getconfno,3)
591 exten = 7,n,GotoIf($[${confno} >= 0 & ${confno} <= 9]?enter)
592 exten = 7,n,Playback(conf-invalid)
593 exten = 7,n,Goto(repeat)
594 exten = 7,n(enter),Playback(conf-placeintoconf)
595 exten = 7,n,SayNumber(${confno})
596 exten = 7,n,Set(SPYGROUP=conference-${confno})
597 exten = 7,n,Set(confopt=cCpsMvio)
598 exten = 7,n,GotoIf($[${confno} >= 4 & ${confno} <= 9]?l1:l2)
599 exten = 7,n(l1),Set(confopt=${confopt}i)
600 exten = 7,n(l2),GotoIf($[${confno} >= 7 & ${confno} <= 9]?l3:l4)
601 exten = 7,n(l3),Set(confopt=${confopt}r)
602 exten = 7,n,Set(MEETME_RECORDINGFILE=${MEETME_SPOOLDIR}/meetme-conference-${confno}-${STRFTIME(${EPOCH},UTC,%Y%m%d%H%M)})
603 exten = 7,n,Set(MEETME_RECORDINGFORMAT=wav49)
604 exten = 7,n,Playback(this-call-may-be-monitored-or-recorded)
605 exten = 7,n(l4),MeetMe(${confno},${confopt})
606 exten = 7,n,Playback(beep)
607 exten = 7,n,Wait(1)
608 exten = 7,n,Playback(vm-goodbye)
609 exten = 7,n,Hangup
611 ; internal service <voicemail> #80/#*80<n>
612 exten = voicemail,1,Goto(80,1)
613 exten = 80,1,GotoIf($[${MAILBOX_EXISTS(${CALLERID(num)}@internal)} == 0]?novm)
614 exten = 80,n,VoiceMailMain(${CALLERID(num)}@internal,s)
615 exten = 80,n,Hangup
616 exten = 80,n(novm),Playback(invalid)
617 exten = 80,n,Hangup
618 exten = _*80.,1,GotoIf($[${MAILBOX_EXISTS(${EXTEN:3}@internal)} == 0]?novm)
619 exten = _*80.,n,VoiceMailMain(${EXTEN:3}@internal)
620 exten = _*80.,n,Hangup
621 exten = _*80.,n(novm),Playback(invalid)
622 exten = _*80.,n,Hangup
624 ; internal service <echo> #81
625 exten = echo,1,Goto(81,1)
626 exten = 81,1,Answer
627 exten = 81,n,Playback(demo-echotest)
628 exten = 81,n,Wait(1)
629 exten = 81,n,Playback(beep)
630 exten = 81,n,Echo
631 exten = 81,n,Wait(1)
632 exten = 81,n,Playback(demo-echodone)
633 exten = 81,n,Wait(1)
634 exten = 81,n,Playback(vm-goodbye)
635 exten = 81,n,Hangup
637 ; internal service <reload> #82
638 exten = reload,1,Goto(82,1)
639 exten = 82,1,Answer
640 exten = 82,n,Read(pin,conf-getpin,4)
641 exten = 82,n,GotoIf($[${pin} = 1234]?ok)
642 exten = 82,n,Playback(conf-invalidpin)
643 exten = 82,n,Hangup
644 exten = 82,n(ok),Playback(beep)
645 exten = 82,n,Wait(1)
646 exten = 82,n,Playback(beep)
647 exten = 82,n,Wait(1)
648 exten = 82,n,Playback(beep)
649 exten = 82,n,Wait(1)
650 exten = 82,n,System(@l_prefix@/sbin/asterisk -rx reload)
651 exten = 82,n,Hangup
653 ; external outgoing ISDN (via SIP-to-ISDN gateway call-through)
654 ;exten = _0.,1,Set(number=${EXTEN:1})
655 ;exten = _0.,n,Set(enum=${ENUMLOOKUP(+${number},ALL)})
656 ;exten = _0.,n,Set(enum_is_sip_url=${REGEX("^SIP/.+" ${enum})})
657 ;exten = _0.,n,GotoIf($["${enum_is_sip_url}" = "1"]?sip:isdn)
658 ;exten = _0.,n(sip),Dial(${enum},60,o)
659 ;exten = _0.,n,Goto(_0.,7)
660 ;exten = _0.,n(isdn),Dial(SIP/gw,60,D(w1234w0#31#${number}#))
661 ;exten = _0.,n,Hangup
663 ; internal outgoing SIP call (part 1/2)
664 ; (notice sort-order trickery!)
665 include = internal-siponly
667 [internal-siponly]
668 ; internal outgoing SIP call (part 2/2)
669 ; (notice sort-order trickery!)
670 exten = _.[@].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},60,o)
671 exten = _.[@].,n,Hangup
672 exten = _.[@].,102,Busy
674 </file>
675 <file name="enum.conf">
676 ;;
677 ;; enum.conf -- Asterisk ENUM configuration
678 ;;
680 [general]
681 search = e164.arpa
682 search = e164.org
684 </file>
685 <file name="musiconhold.conf">
686 ;;
687 ;; musiconhold.conf -- Asterisk music on hold configuration
688 ;;
690 [default]
691 mode = files
692 directory = @l_prefix@/share/asterisk/moh
694 </file>
695 <file name="voicemail.conf">
696 ;;
697 ;; voicemail.conf -- Asterisk voice mail configuration
698 ;;
700 [general]
701 format = wav49
702 serveremail = example@example.com
703 attach = yes
704 maxmsg = 20
705 maxsecs = 180
706 minsecs = 3
707 maxgreet = 60
708 skipms = 3000
709 maxsilence = 10
710 silencethreshold = 128
711 maxlogins = 3
712 charset = ISO-8859-1
713 pbxskip = yes
714 fromstring = Asterisk PBX
715 usedirectory = yes
716 emailsubject = [PBX]: New voice message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
717 emailbody = Dear ${VM_NAME},\n\njust wanted to let you know you were left a ${VM_DUR} long\nvoice message (number ${VM_MSGNUM}) in voice mailbox ${VM_MAILBOX}\nfrom caller ${VM_CALLERID},\non ${VM_DATE}.\nYou might want to check it when you get a chance. Thanks!\n\n\t\t\t\t-- OpenPKG Asterisk PBX\n
718 pagerfromstring = Asterisk PBX
719 pagersubject = New VM
720 pagerbody = New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}
721 emaildateformat = %A, %d %B %Y %H:%M:%S %r
722 mailcmd = @l_prefix@/sbin/sendmail -t
724 [default]
726 [external]
727 1 = 1234,Example,example@example.com,,delete=yes
729 [internal]
730 11 = 1234,Mr. Foo,foo@example.com,,delete=no
731 12 = 1234,Mr. Bar,bar@example.com,,delete=no
733 </file>
734 <file name="cdr.conf">
735 ;;
736 ;; cdr.conf -- Asterisk Call Detail Record (CDR) configuration
737 ;;
739 [general]
740 enable = yes
741 unanswered = no
742 batch = no
743 size = 100
744 time = 300
745 scheduleronly = no
746 safeshutdown = yes
747 endbeforehexten = yes
749 </file>
750 <file name="cdr_custom.conf">
751 ;;
752 ;; cdr_custom.conf -- Asterisk Call Detail Record (CDR) via Comma Separated Value (CSV) format configuration
753 ;;
755 [mappings]
756 master.csv = "${CDR(start)}", "${CDR(answer)}", "${CDR(end)}", "${CDR(duration)}", "${CDR(billsec)}", "${CDR(clid)}", "${CDR(src)}", "${CDR(dst)}", "${CDR(dcontext)}", "${CDR(channel)}", "${CDR(dstchannel)}", "${CDR(lastapp)}", "${CDR(lastdata)}", "${CDR(disposition)}", "${CDR(amaflags)}", "${CDR(accountcode)}", "${CDR(uniqueid)}", "${CDR(userfield)}"
758 </file>
759 <file name="cdr_sqlite3_custom.conf">
760 ;;
761 ;; cdr_sqlite3_custom.conf -- Asterisk Call Detail Record (CDR) via SQLite RDBMS format configuration
762 ;;
764 [master]
765 table = cdr
766 columns = start, answer, end, duration, billsec, clid, src, dst, dcontext, channel, dstchannel, lastapp, lastdata, disposition, amaflags, accountcode, uniqueid, userfield
767 values = "${CDR(start)}", "${CDR(answer)}", "${CDR(end)}", "${CDR(duration)}", "${CDR(billsec)}", "${CDR(clid)}", "${CDR(src)}", "${CDR(dst)}", "${CDR(dcontext)}", "${CDR(channel)}", "${CDR(dstchannel)}", "${CDR(lastapp)}", "${CDR(lastdata)}", "${CDR(disposition)}", "${CDR(amaflags)}", "${CDR(accountcode)}", "${CDR(uniqueid)}", "${CDR(userfield)}"
769 </file>
770 <file name="cdr_manager.conf">
771 ;;
772 ;; cdr_manager.conf -- Asterisk Call Detail Record (CDR) via Asterisk Manager Interface (AMI) configuration
773 ;;
775 [general]
776 enabled = yes
778 </file>
779 <file name="meetme.conf">
780 ;;
781 ;; meetme.conf -- Asterisk conference configuration
782 ;;
784 [general]
785 audiobuffers = 32
786 ;schedule = yes
787 ;logmembercount = yes
788 ;fuzzystart = 300
789 ;earlyalert = 3600
790 ;endalert = 120
792 [rooms]
793 conf = 0
794 conf = 1
795 conf = 2
796 conf = 3
797 conf = 4
798 conf = 5
799 conf = 6
800 conf = 7
801 conf = 8
802 conf = 9,1234,1234
804 </file>
805 <file name="codecs.conf">
806 ;;
807 ;; codecs.conf -- Asterisk codec configuration
808 ;;
810 [speex]
811 quality = 6
812 complexity = 4
813 enhancement = true
814 vad = true
815 vbr = true
816 abr = 8000
817 vbr_quality = 5
818 dtx = false
819 preprocess = false
820 pp_vad = false
821 pp_agc = false
822 pp_agc_level = 8000
823 pp_denoise = false
824 pp_dereverb = false
825 pp_dereverb_decay = 0.4
826 pp_dereverb_level = 0.3
828 [plc]
829 genericplc = true
831 </file>
832 <file name="chan_dahdi.conf">
833 ;;
834 ;; chan_dahdi.conf -- Asterisk DAHDI channel configuration
835 ;;
837 ; (an empty configuration is ok, but required even for DAHDI "dahdidummy" only)
838 [trunkgroups]
839 [channels]
841 </file>
842 <file name="capi.conf">
843 ;;
844 ;; capi.conf -- Asterisk ISDN/CAPI channel configuration
845 ;;
847 [general]
848 nationalprefix = 0
849 internationalprefix= 00
850 rxgain = 1.0
851 txgain = 1.0
852 ulaw = no
853 debug = yes
855 [ISDN1]
856 isdnmode = msn
857 incomingmsn = *
858 controller = 0
859 group = 1
860 ;prefix = 0
861 softdtmf = off
862 relaxdtmf = off
863 accountcode =
864 context = external
865 holdtype = local
866 ;immediate = yes
867 echocancel = no
868 echosquelch = no
869 ;echotail = 64
870 ;bridge = yes
871 ;callgroup = 1
872 ;deflect = 1234567
873 devices = 2
874 ;dtmf_generate = yes
876 </file>
877 <file name="features.conf">
878 ;;
879 ;; features.conf -- Asterisk call features configuration
880 ;;
882 [general]
883 ;parkext = 700
884 ;parkpos = 701-720
885 ;context = parkedcalls
887 </file>
888 <file name="festival.conf">
889 ;;
890 ;; festival.conf -- Asterisk festival configuration
891 ;;
893 [general]
894 host = localhost ; default 'localhost'
895 port = 1314 ; default '1314'
896 usecache = no ; default 'no'
898 ; If usecache=yes, a directory to store waveform cache files.
899 ; The cache is never cleared (yet), so you must take care of cleaning it
900 ; yourself (just delete any or all files from the cache).
901 ; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
902 ; Defaults to /tmp/
903 ;
904 ;cachedir = /opsw/var/asterisk/festivalcache/
905 ;
906 ; Festival command to send to the server.
907 ; Defaults to: (tts_textasterisk "%s" 'file)(quit)\n
908 ; %s is replaced by the desired text to say. The command MUST end with a
909 ; (quit) directive, or the cache handling mechanism will hang. Do not
910 ; forget the \n at the end.
911 ;
912 festivalcommand = (tts_textasterisk "%s" 'file)(quit)\n
914 </file>
915 <file name="gtalk.conf">
916 ;;
917 ;; gtalk.conf -- Asterisk GTalk configuration
918 ;;
920 [general]
921 ;context = default ; Context to dump call into
922 ;bindaddr = 0.0.0.0 ; Address to bind to
923 ;externip = 127.0.0.1 ; Set your external ip if you are behind a NAT.
924 ;stunaddr = <hostname> ; Get your external ip from a STUN server.
925 ; ; Note, if the STUN query is successful, this
926 ; ; will replace any value placed in externip;
927 ;allowguest = yes ; Allow calls from people not in list of peers
929 [guest] ; special account for options on guest account
930 ;disallow = all
931 ;allow = ulaw
932 ;context = guest
934 [ogorman]
935 ;username = <person>@gmail.com ; username of the peer you're
936 ; ; calling or accepting calls from
937 ;disallow = all
938 ;allow = ulaw
939 ;context = default
940 ;connection = asterisk ; client or component in jabber.conf
941 ; ; for the call to leave on.
943 </file>
944 <file name="jabber.conf">
945 ;;
946 ;; jabber.conf -- Asterisk Jabber configuration
947 ;;
949 [general]
950 ;debug = yes
951 ;autoprune = yes
952 ;autoregister = yes
954 ;[asterisk]
955 ;type = client
956 ;serverhost = jabber.example.com
957 ;username = asterisk@example.com/asterisk
958 ;secret = asterisk
959 ;priority = 1
960 ;port = 5222
961 ;usetls = no
962 ;usesasl = no
963 ;buddy = buddy@example.com
964 ;status = available
965 ;timeout = 100
967 </file>
968 <file name="indications.conf">
969 ;;
970 ;; indications.conf -- Asterisk tone indications
971 ;;
973 [general]
974 country = us
976 ; United States
977 ; (according to tones in North America)
978 [us]
979 description = United States (US)
980 ringcadence = 2000,4000
981 dial = 350+440
982 busy = 480+620/500,0/500
983 ring = 440+480/2000,0/4000
984 congestion = 480+620/250,0/250
985 callwaiting = 440/300,0/10000
986 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
987 record = 1400/500,0/15000
988 info = !950/330,!1400/330,!1800/330,0
989 stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
991 ; United Kingdom
992 ; (partly according to BT SIN350)
993 [uk]
994 description = United Kingdom (UK)
995 ringcadence = 400,200,400,2000
996 dial = 350+440
997 busy = 400/375,0/375
998 ring = 400+450/400,0/200,400+450/400,0/2000
999 congestion = 400/400,0/350,400/225,0/525
1000 callwaiting = 400/100,0/4000
1001 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
1002 record = 1400/500,0/60000
1003 info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
1004 stutter = 350+440/750,440/750
1006 ; Germany
1007 ; (according to http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf)
1008 [de]
1009 description = Germany (DE)
1010 ringcadence = 1000,4000
1011 dial = 425
1012 busy = 425/480,0/480
1013 ring = 425/1000,0/4000
1014 congestion = 425/240,0/240
1015 callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
1016 dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
1017 record = 1400/80,0/15000
1018 info = 950/330,1400/330,1800/330,0/1000
1019 stutter = 425+400
1021 </file>
1022 <file name="ccss.conf">
1023 ;;
1024 ;; ccss.conf -- Asterisk Call Completion Supplementary Services configuration
1025 ;;
1027 ; Warning! The CCSS implementation brings several changes to the
1028 ; channel components of Asterisk. To use CCSS, specify the number
1029 ; of maximum requests in this file but do not forget to specify
1030 ; the new CCSS related channel specific options in other config
1031 ; files as well! Some are 'cc_offer_timer', 'ccbs_available_timer',
1032 ; 'cc_agent_policy=never' and many more (in other files.)
1034 [general]
1035 ; There is only a single option that may be defined in this file.
1036 ; The cc_max_requests option is a global limit on the number of
1037 ; CC requests that may be in the Asterisk system at any time.
1038 ;
1039 cc_max_requests = 20
1041 </file>
1042 <file name="res_fax.conf">
1043 ;;
1044 ;; res_fax.conf -- Asterisk fax resource configuration
1045 ;;
1047 [general]
1048 ; Maximum Transmission Rate
1049 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
1050 ; Set this value to the maximum desired transfer rate. Default: 14400
1051 maxrate=14400
1053 ; Minimum Transmission Rate
1054 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
1055 ; Set this value to the minimum desired transfer rate. Default: 2400
1056 minrate=2400
1058 ; Send Progress/Status events to manager session
1059 ; Manager events with 'call' class permissions will receive events indicating the
1060 ; steps to initiate a fax session. Fax completion events are always sent to manager
1061 ; sessions with 'call' class permissions, regardless of the value of this option.
1062 ; Default: no
1063 statusevents=yes
1065 ; modem capabilities
1066 ; Possible values are { v17 | v27 | v29 }
1067 ; Set this value to modify the default modem options. Default: v17,v27,v29
1068 modems=v17,v27,v29
1070 ; Enable/disable T.30 ECM (error correction mode) by default.
1071 ; Default: Enabled
1072 ecm=yes
1074 </file>
1075 <file name="res_odbc.conf">
1076 ;;
1077 ;; res_odbc.conf -- Asterisk ODBC resource configuration
1078 ;;
1080 [ENV]
1082 [asterisk-sqlite]
1083 enabled = no
1084 dsn = asterisk-sqlite
1085 username =
1086 password =
1087 pre-connect = no
1088 sanitysql = SELECT 1
1089 ;idlecheck = 3600
1090 backslash_is_escape= yes
1091 share_connections = yes
1092 limit = 10
1094 </file>
1095 <file name="func_odbc.conf">
1096 ;;
1097 ;; func_odbc.conf -- Asterisk ODBC dialplan function configuration
1098 ;;
1100 ; SQLite-based Asterisk Database Access (random SQL access)
1101 ; Set(<variable_name>=${ASTDB_SQL(SELECT [...])})
1102 ; Set(ASTDB_SQL(UPDATE [...]))
1103 [SQL]
1104 prefix = ASTDB
1105 dsn = asterisk-sqlite
1106 readsql = ${ARG1}
1107 writesql = ${ARG1}
1109 ; SQLite-based Asterisk Database Access (fixed key/value access)
1110 ; Set(<variable_name>=${ASTDB_MAP(<key>)})
1111 ; Set(ASTDB_MAP(<key>)=<value>)
1112 [MAP]
1113 prefix = ASTDB
1114 dsn = asterisk-sqlite
1115 readsql = SELECT val FROM map WHERE key='${SQL_ESC(${ARG1})}'
1116 writesql = UPDATE map SET val='${SQL_ESC(${VAL1})}' WHERE key='${SQL_ESC(${ARG1})}'
1117 escapecommas = no
1119 </file>
1120 <file name="asterisk.pem">
1121 -----BEGIN CERTIFICATE-----
1122 MIIDNjCCAp+gAwIBAgIBATANBgkqhkiG9w0BAQQFADCBqTELMAkGA1UEBhMCWFkx
1123 FTATBgNVBAgTDFNuYWtlIERlc2VydDETMBEGA1UEBxMKU25ha2UgVG93bjEXMBUG
1124 A1UEChMOU25ha2UgT2lsLCBMdGQxHjAcBgNVBAsTFUNlcnRpZmljYXRlIEF1dGhv
1125 cml0eTEVMBMGA1UEAxMMU25ha2UgT2lsIENBMR4wHAYJKoZIhvcNAQkBFg9jYUBz
1126 bmFrZW9pbC5kb20wHhcNOTkxMDIxMTgyMTUxWhcNMDExMDIwMTgyMTUxWjCBpzEL
1127 MAkGA1UEBhMCWFkxFTATBgNVBAgTDFNuYWtlIERlc2VydDETMBEGA1UEBxMKU25h
1128 a2UgVG93bjEXMBUGA1UEChMOU25ha2UgT2lsLCBMdGQxFzAVBgNVBAsTDldlYnNl
1129 cnZlciBUZWFtMRkwFwYDVQQDExB3d3cuc25ha2VvaWwuZG9tMR8wHQYJKoZIhvcN
1130 AQkBFhB3d3dAc25ha2VvaWwuZG9tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB
1131 gQC554Ro+VH0dJONqljPBW+C72MDNGNy9eXnzejXrczsHs3Pc92Vaat6CpIEEGue
1132 yG29xagb1o7Gj2KRgpVYcmdx6tHd2JkFW5BcFVfWXL42PV4rf9ziYon8jWsbK2aE
1133 +L6hCtcbxdbHOGZdSIWZJwc/1Vs70S/7ImW+Zds8YEFiAwIDAQABo24wbDAbBgNV
1134 HREEFDASgRB3d3dAc25ha2VvaWwuZG9tMDoGCWCGSAGG+EIBDQQtFittb2Rfc3Ns
1135 IGdlbmVyYXRlZCBjdXN0b20gc2VydmVyIGNlcnRpZmljYXRlMBEGCWCGSAGG+EIB
1136 AQQEAwIGQDANBgkqhkiG9w0BAQQFAAOBgQB6MRsYGTXUR53/nTkRDQlBdgCcnhy3
1137 hErfmPNl/Or5jWOmuufeIXqCvM6dK7kW/KBboui4pffIKUVafLUMdARVV6BpIGMI
1138 5LmVFK3sgwuJ01v/90hCt4kTWoT8YHbBLtQh7PzWgJoBAY7MJmjSguYCRt91sU4K
1139 s0dfWsdItkw4uQ==
1140 -----END CERTIFICATE-----
1141 -----BEGIN RSA PRIVATE KEY-----
1142 MIICXgIBAAKBgQC554Ro+VH0dJONqljPBW+C72MDNGNy9eXnzejXrczsHs3Pc92V
1143 aat6CpIEEGueyG29xagb1o7Gj2KRgpVYcmdx6tHd2JkFW5BcFVfWXL42PV4rf9zi
1144 Yon8jWsbK2aE+L6hCtcbxdbHOGZdSIWZJwc/1Vs70S/7ImW+Zds8YEFiAwIDAQAB
1145 AoGBAKTvnFGKSkUJnNQGe66I0wunGgCA3W7kbarAzEF2qKYhGlZhJQnn68RmVnAW
1146 pXUFvB+vmtu/+4J9OmWBJsGHFvC9xH32a0PWNr7APjAKrjAD8GWS7Z6BjuxN8QhD
1147 WlFMmpYhYIjT1jt7RNfs2gJGS2Ryu3zutUQGwtUB9Pou03dJAkEA6yttwVINFqQP
1148 utgUZ1JUHrN/rE73FzYsF/CwJp5d3rLHenZzLT0iW+kNDLUw/VpzYxK7bF2Qrt/3
1149 QIUWwm2InQJBAMpe+jhNMJeLDLc3tG3zeithT0mFkuzWWmT2PJgQ0V78UWhw/fSn
1150 Qqnq7KBY/DNjlfhezrozLDD73/ccmha0Ax8CQQCBaBlyOtNm9QqO116K6HvPlRiZ
1151 Wa6QQEgNOG3GInknFZu9ILcKWsywZNLAfmgh0gcSqnkmDWqTQD0PbOz0Ok/lAkEA
1152 g24JrfUbwOASww9PhDUju/a36rTwhhZ0oKt3EP+jKsBOErmHhZP3bKlhQoZoTOu5
1153 Y5QXSMChS7LZcwDFZkdE2wJATRgMbhErif+ZRwt9XJRdCo5Sx6ewyGyxjc5gvUyK
1154 KegHcgru/ZC3pGlujRD2LqxgJNAn5QTdW4LK8xVPFySTYg==
1155 -----END RSA PRIVATE KEY-----
1156 </file>
