;
; Sample ADSI Configuration file
;
[intro]
alignment = center
greeting => Welcome to the
greeting => Asterisk
greeting => Open Source PBX
;
; Voice over Frame Relay (Adtran style)
;
; Configuration file
[interfaces]
;
; Default language
;
;language=en
;
; Lines for which we are the user termination. They accept incoming
; and outgoing calls. We use the default context on the first 8 lines
; used by internal phones.
;
context=default
;user => voice00
;user => voice01
;user => voice02
;user => voice03
;user => voice04
;user => voice05
;user => voice06
;user => voice07
; Calls on 16 and 17 come from the outside world, so they get
; a little bit special treatment
context=remote
;user => voice16
;user => voice17
;
; Next we have lines which we only accept calls on, and typically
; do not send outgoing calls on (i.e. these are where we are the
; network termination)
;
;network => voice08
;network => voice09
;network => voice10
;network => voice11
;network => voice12
[general]
;
; Define whether callbacklogins should be stored in astdb for
; persistence. Persistent logins will be reloaded after
; Asterisk restarts.
;
persistentagents=yes
;
; Agent configuration
;
;
[agents]
;
; Define autologoff times if appropriate. This is how long
; the phone has to ring with no answer before the agent is
; automatically logged off (in seconds)
;
;autologoff=15
;
; Define ackcall to require an acknowledgement by '#' when
; an agent logs in using agentcallbacklogin. Default is "no".
;
;ackcall=no
;
; Define wrapuptime. This is the minimum amount of time when
; after disconnecting before the caller can receive a new call
; note this is in milliseconds.
;
;wrapuptime=5000
;
; Define the default musiconhold for agents
; musiconhold => music_class
;
;musiconhold => default
;
; Define updatecdr. This is whether or not to change the source
; channel in the CDR record for this call to agent/agent_id so
; that we know which agent generates the call
;
;updatecdr=no
;
; Group memberships for agents (may change in mid-file)
;
;group=3
;group=1,2
;group=
;
; --------------------------------------------------
; This section is devoted to recording agent's calls
; The keywords are global to the chan_agent channel driver
;
; Enable recording calls addressed to agents. It's turned off by default.
;recordagentcalls=yes
;
; The format to be used to record the calls: wav, gsm, wav49.
; By default its "wav".
;recordformat=gsm
;
; Insert into CDR userfield a name of the the created recording
; By default it's turned off.
;createlink=yes
;
; The text to be added to the name of the recording. Allows forming a url link.
;urlprefix=http://localhost/calls/
;
; The optional directory to save the conversations in. The default is
; /var/spool/asterisk/monitor
;savecallsin=/var/calls
;
; An optional custom beep sound file to play to always-connected agents.
;custom_beep=beep
;
; --------------------------------------------------
;
; This section contains the agent definitions, in the form:
;
; agent => agentid,agentpassword,name
;
;agent => 1001,4321,Mark Spencer
;agent => 1002,4321,Will Meadows
;
; alarmreceiver.conf
;
; Sample configuration file for the Asterisk alarm receiver application.
;
[general]
;
; Specify a timestamp format for the metadata section of the event files
; Default is %a %b %d, %Y @ %H:%M:%S %Z
timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
;
; Specify a command to execute when the caller hangs up
;
; Default is none
;
;eventcmd = yourprogram -yourargs ...
;
; Specify a spool directory for the event files. This setting is required if you want the app to be useful.
; Event files written to the spool directory will be of the template event-XXXXXX, where XXXXXX is a random
; and unique alphanumeric string.
;
; Default is none, and the events will be dropped on the floor.
;
eventspooldir = /tmp
;
; The alarmreceiver app can either log the events one-at-a-time to individual files in the spool
; directory, or it can store them until the caller disconnects and write them all to one file.
;
; The default setting for logindividualevents is no.
;
logindividualevents = no
;
; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. to 10000 msec. The
; default is 2000 msec. Note: if you wish to test the receiver by entering digits manually, set this
; to a reasonable time out like 10000 milliseconds.
fdtimeout = 2000
;
; The timeout for receiving subsequent DTMF digits is adjustable from 110 msec. to 4000 msec. The
; default is 200 msec. Note: if you wish to test the receiver by entering digits manually, set this
; to a reasonable time out like 4000 milliseconds.
;
sdtimeout = 200
;
; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. The default is 8192
; This shouldn't need to be messed with, but is included just in case there are problems with
; signal levels.
;
loudness = 8192
;
; The db-family setting allows the user to capture statistics on the number of calls, and the errors
; the alarm receiver sees. The default is for no db-family name to be defined and the database logging
; to be turned off.
;
;db-family = yourfamily:
;
; End of alarmreceiver.conf
;
;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console? Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=local
;
; Default extension to call
;
extension=s
;
; Default language
;
;language=en
;
; Silence supression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary. Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; To set which ALSA device to use, change this parameter
;input_device=hw:0,0
;output_device=default
;
; Answering Machine Detection Configuration
;
[general]
initial_silence = 2500 ; Maximum silence duration before the greeting.
; If exceeded then MACHINE.
greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
after_greeting_silence = 800 ; Silence after detecting a greeting.
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
; on a HUMAN or MACHINE
min_word_length = 100 ; Minimum duration of Voice to considered as a word
between_words_silence = 50 ; Minimum duration of silence after a word to consider
; the audio what follows as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded then MACHINE
silence_threshold = 256
[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run
astlogdir => /var/log/asterisk
; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl
;
; Asterisk Call Detail Record engine configuration
;
; CDR is Call Detail Record, which provides logging services via a variety of
; pluggable backend modules. Detailed call information can be recorded to
; databases, files, etc. Useful for billing, fraud prevention, compliance with
; Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more.
;
[general]
; Define whether or not to use CDR logging. Setting this to "no" will override
; any loading of backend CDR modules. Default is "yes".
;enable=yes
; Define the CDR batch mode, where instead of posting the CDR at the end of
; every call, the data will be stored in a buffer to help alleviate load on the
; asterisk server. Default is "no".
;
; WARNING WARNING WARNING
; Use of batch mode may result in data loss after unsafe asterisk termination
; ie. software crash, power failure, kill -9, etc.
; WARNING WARNING WARNING
;
;batch=no
; Define the maximum number of CDRs to accumulate in the buffer before posting
; them to the backend engines. 'batch' must be set to 'yes'. Default is 100.
;size=100
; Define the maximum time to accumulate CDRs in the buffer before posting them
; to the backend engines. If this time limit is reached, then it will post the
; records, regardless of the value defined for 'size'. 'batch' must be set to
; 'yes'. Note that time is in seconds. Default is 300 (5 minutes).
;time=300
; The CDR engine uses the internal asterisk scheduler to determine when to post
; records. Posting can either occure inside the scheduler thread, or a new
; thread can be spawned for the submission of every batch. For small batches,
; it might be acceptable to just use the scheduler thread, so set this to "yes".
; For large batches, say anything over size=10, a new thread is recommended, so
; set this to "no". Default is "no".
;scheduleronly=no
; When shutting down asterisk, you can block until the CDRs are submitted. If
; you don't, then data will likely be lost. You can always check the size of
; the CDR batch buffer with the CLI "cdr status" command. To enable blocking on
; submission of CDR data during asterisk shutdown, set this to "yes". Default
; is "yes".
;safeshutdown=yes
;
; Mappings for custom config file
;
[mappings]
Master.csv => "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
;
; Asterisk Call Management CDR
;
[general]
enabled = yes
;
; Note - if the database server is hosted on the same machine as the
; asterisk server, you can achieve a local Unix socket connection by
; setting hostname=localhost
;
; port and sock are both optional parameters. If hostname is specified
; and is not "localhost", then cdr_mysql will attempt to connect to the
; port specified or use the default port. If hostname is not specified
; or if hostname is "localhost", then cdr_mysql will attempt to connect
; to the socket file specified by sock or otherwise use the default socket
; file.
;
;[global]
;hostname=database.host.name
;dbname=asteriskcdrdb
;table=cdr
;password=password
;user=asteriskcdruser
;port=3306
;sock=/tmp/mysql.sock
;userfield=1
;
; cdr_odbc.conf
;
;[global]
;dsn=MySQL-test
;username=username
;password=password
;loguniqueid=yes
;dispositionstring=yes
;table=cdr ;"cdr" is default table name
;usegmtime=no ; set to "yes" to log in GMT
; Sample Asterisk config file for CDR logging to PostgresSQL
[global]
hostname=localhost
port=5432
dbname=asterisk
password=p0stgr3s
user=postgres
table=cdr ;SQL table where CDRs will be inserted
; Sample Asterisk config file for CDR logging to FreeTDS
;[global]
;hostname=fs.malico.loc
;port=1433
;dbname=MalicoHN
;user=mangUsr
;password=
;charset=BIG5
[speex]
;0-10
quality => 3
;0-10
complexity => 4
; true / false
enhancement => true
; true / false
vad => false
; true / false
vbr => false
; 0 = off, otherwise, target bitrate in bps
abr => 0
;0-10
vbr_quality => 5
; true / false
dtx => 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
genericplc => true
[general]
;enable=yes ; enable creation of managed DNS lookups
; default is 'no'
;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds
; default is 300 (5 minutes);
; DUNDi configuration file
;
; For more information about DUNDi, see http://www.dundi.com
;
;
[general]
;
; The "general" section contains general parameters relating
; to the operation of the dundi client and server.
;
; The first part should be your complete contact information
; should someone else in your peer group need to contact you.
;
;department=Your Department
;organization=Your Company, Inc.
;locality=Your City
;stateprov=ST
;country=US
;email=your@email.com
;phone=+12565551212
;
;
; Specify bind address and port number. Default is
; 4520
;
;bindaddr=0.0.0.0
;port=4520
;
; Our entity identifier (Should generally be the MAC address of the
; machine it's running on. Defaults to the first eth address, but you
; can override it here, as long as you set it to the MAC of *something*
; you own!)
;
;entityid=00:07:E9:3B:76:60
;
; Peers shall cache our query responses for the specified time,
; given in seconds. Default is 3600.
;
;cachetime=3600
;
; This defines the max depth in which to search the DUNDi system.
; Note that the maximum time that we will wait for a response is
; (2000 + 200 * ttl) ms.
;
ttl=32
;
; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set
; to yes, then we cancel the whole thing (that's enough time for one
; retransmission only). This is used to keep things from stalling for a long
; time for a host that is not available, but would be ill advised for bad
; connections. In addition to 'yes' or 'no' you can also specify a number
; of milliseconds. See 'qualify' for individual peers to turn on for just
; a specific peer.
;
autokill=yes
;
; pbx_dundi creates a rotating key called "secret", under the family
; 'secretpath'. The default family is dundi (resulting in
; the key being held at dundi/secret).
;
;secretpath=dundi
;
; The 'storehistory' option (also changeable at runtime with
; 'dundi store history' and 'dundi no store history') will
; cause the DUNDi engine to keep track of the last several
; queries and the amount of time each query took to execute
; for the purpose of tracking slow nodes. This option is
; off by default due to performance impacts.
;
;storehistory=yes
[mappings]
;
; The "mappings" section maps DUNDi contexts
; to contexts on the local asterisk system. Remember
; that numbers that are made available under the e164
; DUNDi context are regulated by the DUNDi General Peering
; Agreement (GPA) if you are a member of the DUNDi E.164
; Peering System.
;
; dundi_context => local_context,weight,tech,dest[,options]]
;
; 'dundi_context' is the name of the context being requested
; within the DUNDi request
;
; 'local_context' is the name of the context on the local system
; in which numbers can be looked up for which responses shall be given.
;
; 'weight' is the weight to use for the responses provided from this
; mapping. The number must be >= 0 and < 60000. Since it is totally
; valid to receive multiple reponses to a query, responses received
; with a lower weight are tried first. Note that the weight has a
; special meaning in the e164 context - see the GPA for more details.
;
; 'tech' is the technology to use (IAX, SIP, H323)
;
; 'dest' is the destination to supply for reaching that number. The
; following variables can be used in the destination string and will
; be automatically substituted:
; ${NUMBER}: The number being requested
; ${IPADDR}: The IP address to connect to
; ${SECRET}: The current rotating secret key to be used
;
; Further options may include:
;
; nounsolicited: No unsolicited calls of any type permitted via this
; route
; nocomunsolicit: No commercial unsolicited calls permitted via
; this route
; residential: This number is known to be a residence
; commercial: This number is known to be a business
; mobile: This number is known to be a mobile phone
; nocomunsolicit: No commercial unsolicited calls permitted via
; this route
; nopartial: Do not search for partial matches
;
; There *must* exist an entry in mappings for DUNDi to respond
; to any request, although it may be empty.
;
;e164 => dundi-e164-canonical,0,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;e164 => dundi-e164-customers,100,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;e164 => dundi-e164-via-pstn,400,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;digexten => default,0,IAX2,guest@lappy/${NUMBER}
;asdf =>
;
;
; The remaining sections represent the peers
; that we fundamentally trust. The section name
; represents the name and optionally at a specific
; DUNDi context if you want the trust to be established
; for only a specific DUNDi context.
;
; inkey - What key they will be authenticating to us with
;
; outkey - What key we use to authenticate to them
;
; host - What their host is
;
; order - What search order to use. May be 'primary', 'secondary',
; 'tertiary' or 'quartiary'. In large systems, it is beneficial
; to only query one up-stream host in order to maximize caching
; value. Adding one with primary and one with secondary gives you
; redundancy without sacraficing performance.
;
; include - Includes this peer when searching a particular context
; for lookup (set "all" to perform all lookups with that
; host. This is also the context in which peers are permitted
; to precache.
;
; noinclude - Disincludes this peer when searching a particular context
; for lookup (set "all" to perform no lookups with that
; host.
;
; permit - Permits this peer to search a given DUNDi context on
; the local system. Set "all" to permit this host to
; lookup all contexts. This is also a context for which
; we will create/forward PRECACHE commands.
;
; deny - Denies this peer to search a given DUNDi context on
; the local system. Set "all" to deny this host to
; lookup all contexts.
;
; model - inbound, outbound, or symmetric for whether we receive
; requests only, transmit requests only, or do both.
;
; precache - Utilize/Permit precaching with this peer (to pre
; cache means to provide an answer when no request
; was made and is used so that machines with few
; routes can push those routes up a to a higher level).
; outgoing means we send precache routes to this peer,
; incoming means we permit this peer to send us
; precache routes. symmetric means we do both.
;
; Note: You cannot mix symmetric/outbound model with symmetric/inbound
; precache, nor can you mix symmetric/inbound model with symmetric/outbound
; precache.
;
;
; The '*' peer is special and matches an unspecified entity
;
;
; Sample Primary e164 DUNDi peer
;
;[00:50:8B:F3:75:BB]
;model = symmetric
;host = 64.215.96.114
;inkey = digium
;outkey = misery
;include = e164
;permit = e164
;qualify = yes
;
; Sample Secondary e164 DUNDi peer
;
;[00:A0:C9:96:92:84]
;model = symmetric
;host = misery.digium.com
;inkey = misery
;outkey = ourkey
;include = e164
;permit = e164
;qualify = yes
;order = secondary
;
; Sample "push mode" downstream host
;
;[00:0C:76:96:75:28]
;model = inbound
;host = dynamic
;precache = inbound
;inkey = littleguy
;outkey = ourkey
;include = e164 ; In this case used only for precaching
;permit = e164
;qualify = yes
;
; Sample "push mode" upstream host
;
;[00:07:E9:3B:76:60]
;model = outbound
;precache = outbound
;host = 216.207.245.34
;register = yes
;inkey = dhcp34
;permit = all ; In this case used only for precaching
;include = all
;qualify = yes
;outkey=foo
;[*]
;
;
; ENUM Configuration for resolving phone numbers over DNS
;
; Sample config for Asterisk
; This file is reloaded at "reload enum" in the CLI
;
[general]
;
; The search list for domains may be customized. Domains are searched
; in the order they are listed here.
;
search => e164.arpa
;
; If you'd like to use the E.164.org public ENUM registery in addition
; to the official e164.arpa one, uncomment the following line
;
;search => e164.org
;
; As there are more H323 drivers available you have to select to which
; drive a H323 URI will map. Default is "H323".
;
h323driver => H323
;
; Static and realtime external configuration
; engine configuration
;
; Please read doc/README.extconfig for basic table
; formatting information.
;
[settings]
;
; Static configuration files:
;
; file.conf => driver,database[,table]
;
; maps a particular configuration file to the given
; database driver, database and table (or uses the
; name of the file as the table if not specified)
;
;uncomment to load queues.conf via the odbc engine.
;
;queues.conf => odbc,asterisk,ast_config
;
; The following files CANNOT be loaded from Realtime storage:
; asterisk.conf
; extconfig.conf (this file)
; logger.conf
;
; Additionally, the following files cannot be loaded from
; Realtime storage unless the storage driver is loaded
; early using 'preload' statements in modules.conf:
; manager.conf
; cdr.conf
; rtp.conf
;
;
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is not specified
;
;example => odbc,asterisk,alttable
;iaxusers => odbc,asterisk
;iaxpeers => odbc,asterisk
;sipusers => odbc,asterisk
;sippeers => odbc,asterisk
;voicemail => odbc,asterisk
;extensions => odbc,asterisk
;queues => odbc,asterisk
;queue_members => odbc,asterisk
[general]
static=yes
writeprotect=no
;autofallthrough=yes
clearglobalvars=no
languaje=es
[globals]
OPERADORA=501
GRUPOTODOS=SIP/sip500&SIP/sip501&SIP/sip502&SIP/sip503&SIP/sip553
GRUPOSOPORTE=SIP/sip500&SIP/sip501
GRUPOSOPORTE=SIP/sip500&SIP/sip501
;CONSOLE=Console/dsp ; Console interface for demo
CONSOLE=Zap/1
;CONSOLE=Phone/phone0
:IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1 ; Trunk interface
SECONDTRUNK=Zap/2-1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
;[context]
;exten => someexten,priority[+offset][(alias)],application(arg1,arg2,...)
;exten => someexten,priority[+offset][(alias)],application,arg1|arg2...
;
; Timing list for includes is
;
; <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9 ; so that dialtone remains even after dialing a 9.
[trunklocal]
; Local eight-digit dialing accessed through trunk interface
ignorepat => 9
exten => _9XXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
; Long distance context accessed through trunk
ignorepat => 9
exten => _901XXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkint]
; International long distance through trunk
ignorepat => 9
exten => _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkcell]
exten => _9044XXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9045XXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkservices]
exten => _90X0,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9066,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9071,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
; Long distance context accessed through trunk interface
ignorepat => 9
exten => _901800XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkfrijol]
exten => _91XX,1,Dial(${SECONDTRUNK}/${EXTEN:${TRUNKMSD}}) ; Conmutador de frijol al minuto
[local]
include => trunklocal
[services]
include => trunkservices
[longdistance]
include => trunkld
[international]
include => trunkint
[celular]
include => trunkcell
[paypercall]
include => trunktollfree
[frijolalminuto]
include => trunkfrijol
; ### ### ### ### ### ### ### ### ### ### ### ###
[ptrunklocal]
; Local eight-digit dialing accessed through trunk interface
ignorepat => 9
exten => _9XXXXXXXX,1,Authenticate(123)
exten => _9XXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[ptrunkld]
; Long distance context accessed through trunk
ignorepat => 9
exten => _901XXXXXXXXXX,1,Authenticate(123)
exten => _901XXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[ptrunkint]
; International long distance through trunk
ignorepat => 9
exten => _9001.,1,Authenticate(123)
exten => _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[ptrunkcell]
exten => _9044XXXXXXXXXX,1,Authenticate(123)
exten => _9044XXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9045XXXXXXXXXX,1,Authenticate(123)
exten => _9045XXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[ptrunkservices]
exten => _90X0,1,Authenticate(123)
exten => _90X0,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9066,1,Authenticate(123)
exten => _9066,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[ptrunktollfree]
; Long distance context accessed through trunk interface
ignorepat => 9
exten => _901800XXXXXXX,1,Authenticate(123)
exten => _901800XXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901800NXXXXXX,1,Authenticate(123)
exten => _901800NXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901888NXXXXXX,1,Authenticate(123)
exten => _901888NXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901877NXXXXXX,1,Authenticate(123)
exten => _901877NXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _901866NXXXXXX,1,Authenticate(123)
exten => _901866NXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[ptrunkfrijol]
exten => _91XX,1,Authenticate(123)
exten => _91XX,n,Dial(${SECONDTRUNK}/${EXTEN:${TRUNKMSD}}) ; Conmutador de frijol al minuto
[plocal]
include => ptrunklocal
[pservices]
include => ptrunkservices
[plongdistance]
include => ptrunkld
[pinternational]
include => ptrunkint
[pcelular]
include => ptrunkcell
[ppaypercall]
include => ptrunktollfree
[pfrijolalminuto]
include => ptrunkfrijol
; ### ### ### ### ### ### ### ### ### ### ### ###
; GRUPOS DE TELEFONOS
[oficina]
include => default
include => local
include => services
include => longdistance
include => celular
; include => paypercall
include => frijolalminuto
[macro-stdexten];
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,noop(* - # - * - # - * - # - dial #1)
exten => s,n,Dial(${ARG2},10,rt) ; Ring the interface, 20 seconds maximum
exten => s,n,Set(CHANNEL(language)=es)
exten => _0,1,NoOp,"Extension estandard al numero ${ARG2}"
exten => s,n,NoOp(stdexten ARG1 ${ARG1} ARG2 ${ARG2})
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[macro-alaoperadora]
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,noop(* - # - * - # - * - # - dial #2)
exten => s,n,Dial(${ARG2},20,rt) ; Ring the interface, 20 seconds maximum
exten => s,n,Set(CHANNEL(language)=es)
exten => _0,1,NoOp,"Extension estandard al numero ${ARG2}"
exten => s,n,NoOp(stdexten ARG1 ${ARG1} ARG2 ${ARG2})
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,noop(* - # - * - # - * - # - dial #3)
exten => s-NOANSWER,n,Dial(${GRUPOTODOS},20,rt) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,noop(* - # - * - # - * - # - dial #4)
exten => s-BUSY,n,Dial(${GRUPOTODOS},20,rt) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,noop(* - # - * - # - * - # - dial #5)
exten => a,1,Dial(${GRUPOTODOS},20,rt) ; If they press *, send the user into VoicemailMain
[macro-options]
exten => s,1,NoOp(ejecutando macro options con argumento ${ARG1})
;exten => s,n,Set(CHANNEL(language)=es)
exten => s,n,Wait(1)
exten => s,n,Background(to-reach&CEO&vm-press&digits/500)
exten => s,n,Wait(1)
exten => s,n,Background(to-reach&support&vm-press&digits/555)
exten => s,n,Wait(1)
exten => s,n,Background(to-reach&telephony&vm-press&digits/500&digits/2)
exten => s,n,Wait(2)
exten => s,n,Goto(default,s,1)
[macro-hablaratodos]
exten => s,1,noop(* - # - * - # - * - # - dial #6)
exten => s,n,Dial(${GRUPOTODOS},10,rt) ; Ring the interface, 20 seconds maximum
exten => s,n,Set(CHANNEL(language)=es)
exten => s,n,NoOp,(Extension estandard al numero ${ARG2})
exten => s,n,NoOp(stdexten ARG1 ${ARG1} ARG2 ${ARG2})
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Goto(default,s,1)
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Goto(default,s,1)
exten => s-BUSY,2,Goto(default,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
exten => a,1,Goto(default,s,1) ;
[macro-invalid]
exten => s,1,NoOp(*# *# *#* #* INVALID CATCHER (#${ARG1}) *# *# *#* #*)
exten => s,n,Set(CHANNEL(language)=es)
exten => s,n,Background(pbx-invalid)
exten => s,n,Goto(default,s,restart)
; 004a
[default]
exten => s,1(restart),NoOp,"*# *# *#* #* Operadora / Menu Principal *# *# *#* #*"
exten => s,n,NoOp(DIALSTATUS es '${DIALSTATUS}')
exten => s,n,Noop(ACCOUNTCODE=${ACCOUNTCODE})
exten => s,n,Noop(ANSWEREDTIME=${ANSWEREDTIME})
exten => s,n,Noop(BLINDTRANSFER=${BLINDTRANSFER})
exten => s,n,Noop(CALLERID=${CALLERID(all)})
exten => s,n,Noop(CALLERID(name)=${CALLERID(name)})
exten => s,n,Noop(CALLERID(number)=${CALLERID(number)})
exten => s,n,Noop(CALLINGPRES=${CALLINGPRES})
exten => s,n,Noop(CHANNEL=${CHANNEL})
exten => s,n,Noop(CONTEXT=${CONTEXT})
exten => s,n,Noop(DATETIME=${DATETIME})
exten => s,n,Noop(DIALEDPEERNAME=${DIALEDPEERNAME})
exten => s,n,Noop(DIALEDPEERNUMBER=${DIALEDPEERNUMBER})
exten => s,n,Noop(DIALEDTIME=${DIALEDTIME})
exten => s,n,Noop(DIALSTATUS=${DIALSTATUS})
exten => s,n,Noop(DNID=${DNID})
exten => s,n,Noop(EPOCH=${EPOCH})
exten => s,n,Noop(EXTEN=${EXTEN})
exten => s,n,Noop(HANGUPCAUSE=${HANGUPCAUSE})
exten => s,n,Noop(INVALID_EXTEN=${INVALID_EXTEN})
exten => s,n,Noop(LANGUAGE=${LANGUAGE})
exten => s,n,Noop(MEETMESECS=${MEETMESECS})
exten => s,n,Noop(PRIORITY=${PRIORITY})
exten => s,n,Noop(RDNIS=${RDNIS})
exten => s,n,Noop(SIPDOMAIN=${SIPDOMAIN})
exten => s,n,Noop(SIP_CODEC=${SIP_CODEC})
exten => s,n,Noop(SIPCALLID=${SIPCALLID})
exten => s,n,Noop(SIPUSERAGENT=${SIPUSERAGENT})
exten => s,n,Noop(TIMESTAMP=${TIMESTAMP})
exten => s,n,Noop(TXTCIDNAME=${TXTCIDNAME})
exten => s,n,Noop(UNIQUEID=${UNIQUEID})
exten => s,n,Noop(TOUCH_MONITOR=${TOUCH_MONITOR})
exten => s,n,Noop(MACRO_CONTEXT=${MACRO_CONTEXT})
exten => s,n,Noop(MACRO_EXTEN=${MACRO_EXTEN})
exten => s,n,Noop(MACRO_PRIORITY=${MACRO_PRIORITY})
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(language)=es)
exten => s,n,Wait(1)
exten => s,n(estaeningles),Answer()
exten => s,n,Wait(1)
exten => s,n,Background(if-u-know-ext-dial&vm-reachoper&for-more-options&vm-press&digits/1)
exten => s,n,WaitExten(5) ; Wait for an extension to be dialed.
exten => s,n(reintentar),noop(* - # - * - # - * - # - dial #7)
exten => s,n,Dial(SIP/sip${OPERADORA},10,rt) ; Ring the interface, 20 seconds maximum
;exten => s,n,GoTo(default,s,restart)
exten => s,n,NoOp,"Extension al numero de todos ${GRUPOTODOS}"
exten => s,n,Goto(s-${DIALSTATUS},1) ; jump on (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Wait(1)
exten => s-NOANSWER,1,Background(busy-pls-hold) ; If unavailable, ...
exten => s-NOANSWER,1,Wait(1)
;exten => s-NOANSWER,n,Goto(default,s,restart) ; If they press #, return to start
exten => s-NOANSWER,n,Macro(hablaratodos)
exten => s-BUSY,1,Wait(1) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,Background(busy-pls-hold&vm-starmain) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,Wait(1) ; If busy, send to voicemail w/ busy announce
;exten => s-BUSY,n,Goto(default,s,restart) ; If they press #, return to start
exten => s-BUSY,n,Macro(hablaratodos)
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,Goto(default,s,restart) ; If they press *, send the user into VoicemailMain
;exten => i,1,NoOp(*# *# *#* #* INVALID EXTENSION VAR "i" *# *# *#* #*)
;exten => i,n,Set(CHANNEL(language)=es)
;exten => i,n,Background(pbx-invalid)
;exten => i,n,Playback(pbx-invalid)
;exten => i,n,Goto(default,s,restart)
exten => i,1,Macro(invalid,i)
exten => t,1,NoOp(*# *# *#* #* Extension t ejecutada *# *# *#* #*)
exten => t,n,noop(* - # - * - # - * - # - dial #8)
exten => t,n,Dial(SIP/sip${OPERADORA},10,rt)
exten => 0,1,NoOp(*# *# *#* #* Extension 0 ejecutada *# *# *#* #*)
exten => 0,n,Macro(alaoperadora,${OPERADORA},SIP/sip${OPERADORA}) ; Marca CERO si quieres OPERADORA, en este caso es un telefono fisico, o no sirve porque se va al inicio del dp
exten => 1,1,NoOp(*# *# *#* #* Extension 1 ejecutada *# *# *#* #*)
exten => 1,n,Macro(options,1) ; Escuchar mas opciones
exten => 500,1,Macro(stdexten,500,SIP/sip500) ; tel Nombre
exten => 501,1,Macro(stdexten,501,SIP/sip501) ; tel MEMO
exten => 502,1,Macro(stdexten,502,SIP/sip502) ; tel Nombre
exten => 503,1,Macro(stdexten,503,SIP/sip503) ; tel Nombre
exten => 550,1,Macro(stdexten,550,SIP/sip550) ; tel Nombre
exten => 553,1,Macro(stdexten,553,SIP/sip553) ; xten Nombre
exten => 555,1,Macro(hablaratodos,${GRUPOSOPORTE}) ; SOPORTE
exten => 700,1,NoOp(Marcan 700, vamos a la operadora!)
exten => 700,n,Goto(default,s,restart) ; Marca 700 si quieres que te conteste el inicio
exten => 8500,1,NoOp(Le hablan al VoiceMailMan! uorales!) ; correo de voz
exten => 8500,n,VoicemailMain ; correo de voz
exten => 8500.,n,GoTo(123469,1) ; correo de voz
exten => X,1,Macro(invalid,1)
exten => X.,1,Macro(invalid,2)
exten => 8,1,Macro(clave1,500)
[macro-clave1]
exten => s,1,Set(CHANNEL(language)=es)
exten => s,n,Authenticate(${ARG1})
exten => s,n,NoOp(DIALSTATUS is '${DIALSTATUS}')
exten => s,n,Background(welcome-mailinator-menu)
exten => s,n(congrats),Background(congrats)
exten => s,n,Wait(3)
exten => s,n,GoTo(s,congrats)
exten => t,1,Goto(default,s,1)
exten => i,1,Goto(default,s,1)
exten => 777,1,noop(* - # - * - # - * - # - dial #10)
exten => 777,n,Dial(IAX2/REMOTE_SERVER/0)
exten => 777,n,Hangup
[macro-mailinatordemo]
exten => s,1,Set(CHANNEL(language)=es)
exten => s,n,Background(congrats)
exten => s,n,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
exten => 1,1,Goto(telemarket-charity,s,1)
exten => 2,1,Goto(telemarket-political,s,1)
exten => 3,1,Goto(telemarket-pollster,s,1)
exten => 4,1,Goto(telemarket-research,s,1)
exten => 5,1,Goto(telemarket-magazine,s,1)
exten => 6,1,Goto(telemarket-commercial,s,1)
exten => 7,1,Goto(telemarket-other,s,1)
; Bienvenido a mailinator IT Services,
; for english press one 1, att-for-language-dial-1.wav
; Si conoce el numero de extension marquelo ahora. att-for-ext-dial-now.wav
; Para asistencia personal marque 0 att-for-personal-dial-0.wav
; si requiere soporte tecnico marque 2, att-for-support-dial-2.wav
; para ventas marque 3. att-for-sales-dial-3.wav
; Gracias att-thank-you.wav
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
;xfersound = beep ; to indicate an attended transfer is complete
;xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the 'next' parking space. Defaults to 'first' available
;pickupexten = *8 ; Configure the pickup extension. Default is *8
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
;blindxfer => #1 ; Blind transfer
;disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
;atxfer => *2 ; Attended transfer
;
; Festival Configuration
;
[general]
;
; Host which runs the festival server (default : localhost);
;
;host=localhost
;
; Port on host where the festival server runs (default : 1314)
;
;port=1314
;
; Use cache (yes, no - defaults to no)
;
;usecache=yes
;
; If usecache=yes, a directory to store waveform cache files.
; The cache is never cleared (yet), so you must take care of cleaning it
; yourself (just delete any or all files from the cache).
; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
; Defaults to /tmp/
;
;cachedir=/var/lib/asterisk/festivalcache/
;
; Festival command to send to the server.
; Defaults to: (tts_textasterisk "%s" 'file)(quit)\n
; %s is replaced by the desired text to say. The command MUST end with a
; (quit) directive, or the cache handling mechanism will hang. Do not
; forget the \n at the end.
;
;festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
;
;
; Find-Me / Follow-Me Configuration File
[general]
;
featuredigittimeout=>5000
; The number of ms to wait for a digit input for the callee on whether to take the call or
; not before we consider them "done" entering digits.
;
takecall=>1
; The global default keypress for the callee to take taking the current call. This can be
; a single digit or multiple digits. Default is "1".
;
declinecall=>2
; The global default keypress for the callee to decline taking the current call. This can
; be a single digit or multiple digits. Default is "2".
;
call-from-prompt=>followme/call-from
; The global default for the 'Incoming call from' message.
;
norecording-prompt=>followme/no-recording
; The global default for the 'You have an incoming call' message when the caller elects
; not to leave their name or the option isn't set for them to do so.
;
options-prompt=>followme/options
; The global default for the 'Press 1 to accept this call or press 2 to decline it' message.
;
pls-hold-prompt=>followme/pls-hold-while-try
; The global default for 'Please hold while we try and connect your call' message.
;
status-prompt=>followme/status
; The global default for 'The party you're calling isn't at their desk' message.
;
sorry-prompt=>followme/sorry
; The global default for 'I'm sorry, but we were unable to locate your party' message.
;
;
[default]
musicclass=>default
; The moh class that should be used for the caller while they are waiting to be connected.
context=>default
; The context to dial the numbers from
number=>01233456,25
; The a follow-me number to call. The format is:
; number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ]
; You can specify as many of these numbers as you like. They will be dialed in the
; order that you specify them in the config file OR as specified with the order field
; on the number prompt. As you can see from the example, forked dialing of multiple
; numbers in the same step is supported with this application if you'd like to dial
; multiple numbers in the same followme step.
; It's also important to note that the timeout value is not the same
; as the timeout value you would use in app_dial. This timeout value is the amount of
; time allowed between the time the dialing step starts and the callee makes a choice
; on whether to take the call or not. That being the case, you may want to account for
; this time, and make this timeout longer than a timeout you might specify in app_dial.
takecall=>1
; The keypress for the callee to take taking the current call. This can be
; a single digit or multiple digits. Default is the global default.
;
declinecall=>2
; The keypress for the callee to decline taking the current call. This can
; be a single digit or multiple digits. Default is the global default.
;
call-from-prompt=>followme/call-from
; The 'Incoming call from' message prompt. Default is the global default.
;
followme-norecording-prompt=>followme/no-recording
; The 'You have an incoming call' message prompt when the caller elects
; not to leave their name or the option isn't set for them to do so. Default
; is the global default.
;
followme-options-prompt=>followme/followme-options
; The 'Press 1 to accept this call or press 2 to decline it' message prompt.
; Default is the global default.
;
followme-pls-hold-prompt=>followme/pls-hold-while-try
; The 'Please hold while we try and connect your call' message prompt.
; Default is the global default.
;
followme-status-prompt=>followme/followme-status
; The 'The party you're calling isn't at their desk' message prompt.
; Default is the global default.
;
followme-sorry-prompt=>followme/followme-sorry
; The 'I'm sorry, but we were unable to locate your party' message prompt. Default
; is the global default.
;
; func_odbc.conf
;
; Each context is a separately defined function. By convention, all
; functions are entirely uppercase, so the defined contexts should also
; be all-uppercase, but there is nothing that enforces this. All functions
; are case-sensitive, however.
;
; For substitution, you have ${ARG1}, ${ARG2} ... ${ARGn}
; for the arguments to each SQL statement.
;
; In addition, for write statements, you have ${VAL1}, ${VAL2} ... ${VALn}
; parsed, just like arguments, for the values. In addition, if you want the
; whole value, never mind the parsing, you can get that with ${VALUE}.
;
;
; If you have data which may potentially contain single ticks, you may wish
; to use the dialplan function SQL_ESC() to escape the data prior to its
; inclusion in the SQL statement.
; ODBC_SQL - Allow an SQL statement to be built entirely in the dialplan
[SQL]
dsn=mysql1
read=${ARG1}
; ODBC_ANTIGF - A blacklist.
[ANTIGF]
dsn=mysql1
read=SELECT COUNT(*) FROM exgirlfriends WHERE callerid='${SQL_ESC(${ARG1})}'
; ODBC_PRESENCE - Retrieve and update presence
[PRESENCE]
dsn=mysql1
read=SELECT location FROM presence WHERE id='${SQL_ESC(${ARG1})}'
write=UPDATE presence SET location='${SQL_ESC(${VAL1})}' WHERE id='${SQL_ESC(${ARG1})}'
;prefix=OFFICE ; Changes this function from ODBC_PRESENCE to OFFICE_PRESENCE
;escapecommas=no ; Normally, commas within a field are escaped such that each
; field may be separated into individual variables with ARRAY.
; This option turns that behavior off [default=yes].
;[general]
;context=default ;;Context to dump call into
;allowguest=yes ;;Allow calls from people not in
;;list of peers
;
;[guest] ;;special account for options on guest account
;disallow=all
;allow=ulaw
;context=guest
;
;[ogorman]
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;disallow=all
;allow=ulaw
;context=default
;connection=asterisk ;;client or component in jabber.conf
;;for the call to leave on.
;
; The NuFone Network's
; Open H.323 driver configuration
;
[general]
port = 1720
;bindaddr = 1.2.3.4 ; this SHALL contain a single, valid IP address for this machine
;tos=lowdelay
;
; You may specify a global default AMA flag for iaxtel calls. It must be
; one of 'default', 'omit', 'billing', or 'documentation'. These flags
; are used in the generation of call detail records.
;
;amaflags = default
;
; You may specify a default account for Call Detail Records in addition
; to specifying on a per-user basis
;
;accountcode=lss0101
;
; You can fine tune codecs here using "allow" and "disallow" clauses
; with specific codecs. Use "all" to represent all formats.
;
;disallow=all
;allow=all ; turns on all installed codecs
;disallow=g723.1 ; Hm... Proprietary, don't use it...
;allow=gsm ; Always allow GSM, it's cool :)
;allow=ulaw ; see doc/rtp-packetization for framing options
;
; User-Input Mode (DTMF)
;
; valid entries are: rfc2833, inband
; default is rfc2833
;dtmfmode=rfc2833
;
; Default RTP Payload to send RFC2833 DTMF on. This is used to
; interoperate with broken gateways which cannot successfully
; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
;
; You may also specify on either a per-peer or per-user basis below.
;dtmfcodec=101
;
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
; DISABLE - Disable the use of a GK
; <IP address> or <Host name> - The acutal IP address or hostname of your GK
;gatekeeper = DISABLE
;
;
; Tell Asterisk whether or not to accept Gatekeeper
; routed calls or not. Normally this should always
; be set to yes, unless you want to have finer control
; over which users are allowed access to Asterisk.
; Default: YES
;
;AllowGKRouted = yes
;
; When the channel works without gatekeeper, there is possible to
; reject calls from anonymous (not listed in users) callers.
; Default is to allow anonymous calls.
;
;AcceptAnonymous = yes
;
; Optionally you can determine a user by Source IP versus its H.323 alias.
; Default behavour is to determine user by H.323 alias.
;
;UserByAlias=no
;
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; use user authentication at all.
;
;context=default
;
; Use this option to help Cisco (or other) gateways to setup backward voice
; path to pass inband tones to calling user (see, for example,
; http://www.cisco.com/warp/public/788/voip/ringback.html)
;
; Add PROGRESS information element to SETUP message sent on outbound calls
; to notify about required backward voice path. Valid values are:
; 0 - don't add PROGRESS information element (default);
; 1 - call is not end-end ISDN, further call progress information can
; possibly be available in-band;
; 3 - origination address is non-ISDN (Cisco accepts this value only);
; 8 - in-band information or an appropriate pattern is now available;
;progress_setup = 3
;
; Add PROGRESS information element (IE) to ALERT message sent on incoming
; calls to notify about required backwared voice path. Valid values are:
; 0 - don't add PROGRESS IE (default);
; 8 - in-band information or an appropriate pattern is now available;
;progress_alert = 8
;
; Generate PROGRESS message when H.323 audio path has established to create
; backward audio path at other end of a call.
;progress_audio = yes
;
; Specify how to inject non-standard information into H.323 messages. When
; the channel receives messages with tunneled information, it automatically
; enables the same option for all further outgoing messages independedly on
; options has been set by the configuration. This behavior is required, for
; example, for Cisco CallManager when Q.SIG tunneling is enabled for a
; gateway where Asterisk lives.
; The option can be used multiple times, one option per line.
;tunneling=none ; Totally disable tunneling (default)
;tunneling=cisco ; Enable Cisco-specific tunneling
;tunneling=qsig ; Enable tunneling via Q.SIG messages
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; H323 channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The H323 channel can accept jitter,
; thus an enabled jitterbuffer on the receive H323 side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a H323
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
;
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is one.
;
; Example: if someone calls time@your.asterisk.box.com
; Asterisk will send the call to the extension 'time'
; in the context default
;
; [default]
; exten => time,1,Answer
; exten => time,2,Playback,current-time
;
; Keyword's 'prefix' and 'e164' are only make sense when
; used with a gatekeeper. You can specify either a prefix
; or E.164 this endpoint is responsible for terminating.
;
; Example: The H.323 alias 'det-gw' will tell the gatekeeper
; to route any call with the prefix 1248 to this alias. Keyword
; e164 is used when you want to specifiy a full telephone
; number. So a call to the number 18102341212 would be
; routed to the H.323 alias 'time'.
;
;[time]
;type=h323
;e164=18102341212
;context=default
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
;
; Inbound H.323 calls from BillyBob would land in the incoming
; context with a maximum of 4 concurrent incoming calls
;
;
; Note: If keyword 'incominglimit' are omitted Asterisk will not
; enforce any maximum number of concurrent calls.
;
;[BillyBob]
;type=user
;host=192.168.1.1
;context=incoming
;incominglimit=4
;h245Tunneling=no
;
;
; Outbound H.323 call to Larry using SlowStart
;
;[Larry]
;type=peer
;host=192.168.2.1
;fastStart=no
;
; Asterisk Builtin mini-HTTP server
;
;
[general]
;
; Whether HTTP interface is enabled or not. Default is no.
;
;enabled=yes
;
; Whether Asterisk should serve static content from http-static
; Default is no.
;
;enablestatic=yes
;
; Address to bind to. Default is 0.0.0.0
;
bindaddr=127.0.0.1
;
; Port to bind to (default is 8088)
;
bindport=8088
;
; Prefix allows you to specify a prefix for all requests
; to the server. The default is "asterisk" so that all
; requests must begin with /asterisk
;
;prefix=asterisk
; The post_mappings section maps URLs to real paths on the filesystem. If a
; POST is done from within an authenticated manager session to one of the
; configured POST mappings, then any files in the POST will be placed in the
; configured directory.
;
;[post_mappings]
;
; In this example, if the prefix option is set to "asterisk", then using the
; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
;uploads = /var/lib/asterisk/uploads/
;
; Inter-Asterisk eXchange driver definition
;
; This configuration is re-read at reload
; or with the CLI command
; reload chan_iax2.so
;
; General settings, like port number to bind to, and
; an option address (the default is to bind to all
; local addresses).
;
[general]
;bindport=4569 ; bindport and bindaddr may be specified
; ; NOTE: bindport must be specified BEFORE bindaddr
; ; or may be specified on a specific bindaddr if followed by
; ; colon and port (e.g. bindaddr=192.168.0.1:4569)
;bindaddr=192.168.0.1 ; more than once to bind to multiple
; ; addresses, but the first will be the
; ; default
;
; Set iaxcompat to yes if you plan to use layered switches or
; some other scenario which may cause some delay when doing a
; lookup in the dialplan. It incurs a small performance hit to
; enable it. This option causes Asterisk to spawn a separate thread
; when it receives an IAX DPREQ (Dialplan Request) instead of
; blocking while it waits for a response.
;
;iaxcompat=yes
;
; Disable UDP checksums (if nochecksums is set, then no checkums will
; be calculated/checked on systems supporting this feature)
;
;nochecksums=no
;
;
; For increased security against brute force password attacks
; enable "delayreject" which will delay the sending of authentication
; reject for REGREQ or AUTHREP if there is a password.
;
;delayreject=yes
;
; You may specify a global default AMA flag for iaxtel calls. It must be
; one of 'default', 'omit', 'billing', or 'documentation'. These flags
; are used in the generation of call detail records.
;
;amaflags=default
;
; You may specify a default account for Call Detail Records in addition
; to specifying on a per-user basis
;
;accountcode=lss0101
;
; You may specify a global default language for users.
; Can be specified also on a per-user basis
; If omitted, will fallback to english
;
;language=en
;
; Specify bandwidth of low, medium, or high to control which codecs are used
; in general.
;
bandwidth=low
;
; You can also fine tune codecs here using "allow" and "disallow" clauses
; with specific codecs. Use "all" to represent all formats.
;
;allow=all ; same as bandwidth=high
;disallow=g723.1 ; Hm... Proprietary, don't use it...
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
;allow=gsm ; Always allow GSM, it's cool :)
;
; You can adjust several parameters relating to the jitter buffer.
; The jitter buffer's function is to compensate for varying
; network delay.
;
; There are presently two jitterbuffer implementations available for * and chan_iax2;
; the classic and the new, channel/application independent implementation. These
; are controlled at compile-time. The new jitterbuffer additionally has support for PLC
; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost
; packets.
;
; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio
; will be dejittered by the jitter buffer at the other end.
;
; jitterbuffer=yes|no: global default as to whether you want
; the jitter buffer at all.
;
; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
; we don't want to do jitterbuffering on the switch, since the endpoints
; can each handle this. However, some endpoints may have poor jitterbuffers
; themselves, so this option will force * to always jitterbuffer, even in this case.
; [This option presently applies only to the new jitterbuffer implementation]
;
; dropcount: the jitter buffer is sized such that no more than "dropcount"
; frames would have been "too late" over the last 2 seconds.
; Set to a small number. "3" represents 1.5% of frames dropped
; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
;
; maxjitterbuffer: a maximum size for the jitter buffer.
; Setting a reasonable maximum here will prevent the call delay
; from rising to silly values in extreme situations; you'll hear
; SOMETHING, even though it will be jittery.
;
; resyncthreshold: when the jitterbuffer notices a significant change in delay
; that continues over a few frames, it will resync, assuming that the change in
; delay was caused by a timestamping mix-up. The threshold for noticing a change
; in delay is measured as twice the measured jitter plus this resync threshold.
; Resycning can be disabled by setting this parameter to -1.
; [This option presently applies only to the new jitterbuffer implementation]
;
; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer should
; return in a row. Since some clients do not send CNG/DTX frames to indicate
; silence, the jitterbuffer will assume silence has begun after returning this
; many interpolations. This prevents interpolating throughout a long silence.
; [This option presently applies only to the new jitterbuffer implementation]
;
; maxexcessbuffer: If conditions improve after a period of high jitter,
; the jitter buffer can end up bigger than necessary. If it ends up
; more than "maxexcessbuffer" bigger than needed, Asterisk will start
; gradually decreasing the amount of jitter buffering.
; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
;
; minexcessbuffer: Sets a desired mimimum amount of headroom in
; the jitter buffer. If Asterisk has less headroom than this, then
; it will start gradually increasing the amount of jitter buffering.
; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
;
; jittershrinkrate: when the jitter buffer is being gradually shrunk
; (or enlarged), how many millisecs shall we take off per 20ms frame
; received? Use a small number, or you will be able to hear it
; changing. An example: if you set this to 2, then the jitter buffer
; size will change by 100 millisecs per second.
; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
jitterbuffer=no
forcejitterbuffer=no
;dropcount=2
;maxjitterbuffer=1000
;maxjitterinterps=10
;resyncthreshold=1000
;maxexcessbuffer=80
;minexcessbuffer=10
;jittershrinkrate=1
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
; Should we send timestamps for the individual sub-frames within trunk frames?
; There is a small bandwidth use for these (less than 1kbps/call), but they ensure
; that frame timestamps get sent end-to-end properly. If both ends of all your trunks
; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you
; can probably suppress these. The receiver must also support this feature, although
; they do not also need to have it enabled.
;
; trunktimestamps=yes
;
;
; We can register with another IAX server to let him know where we are
; in case we have a dynamic IP address for example
;
; Register with tormenta using username marko and password secretpass
;
;register => marko:secretpass@tormenta.linux-support.net
;
; Register joe at remote host with no password
;
;register => joe@remotehost:5656
;
; Register marko at tormenta.linux-support.net using RSA key "torkey"
;
;register => marko:[torkey]@tormenta.linux-support.net
;
; Sample Registration for iaxtel
;
; Visit http://www.iaxtel.com to register with iaxtel. Replace "user"
; and "pass" with your username and password for iaxtel. Incoming
; calls arrive at the "s" extension of "default" context.
;
;register => user:pass@iaxtel.com
;
; Sample Registration for IAX + FWD
;
; To register using IAX with FWD, it must be enabled by visiting the URL
; http://www.fwdnet.net/index.php?section_id=112
;
; Note that you need an extension in you default context which matches
; your free world dialup number. Please replace "FWDNumber" with your
; FWD number and "passwd" with your password.
;
;register => FWDNumber:passwd@iax.fwdnet.net
;
;
; You can disable authentication debugging to reduce the amount of
; debugging traffic.
;
;authdebug=no
;
; Finally, you can set values for your TOS bits to help improve
; performance. Valid values are:
; lowdelay -- Minimize delay
; throughput -- Maximize throughput
; reliability -- Maximize reliability
; mincost -- Minimize cost
; none -- No flags
;
tos=lowdelay
;
; If mailboxdetail is set to "yes", the user receives
; the actual new/old message counts, not just a yes/no
; as to whether they have messages. this can be set on
; a per-peer basis as well
;
;mailboxdetail=yes
;
; If regcontext is specified, Asterisk will dynamically
; create and destroy a NoOp priority 1 extension for a given
; peer who registers or unregisters with us. The actual extension
; is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided. More than one regexten may be supplied
; if they are separated by '&'. Patterns may be used in regexten.
;
;regcontext=iaxregistrations
;
; If we don't get ACK to our NEW within 2000ms, and autokill is set
; to yes, then we cancel the whole thing (that's enough time for one
; retransmission only). This is used to keep things from stalling for a long
; time for a host that is not available, but would be ill advised for bad
; connections. In addition to 'yes' or 'no' you can also specify a number
; of milliseconds. See 'qualify' for individual peers to turn on for just
; a specific peer.
;
autokill=yes
;
; codecpriority controls the codec negotiation of an inbound IAX call.
; This option is inherited to all user entities. It can also be defined
; in each user entity separately which will override the setting in general.
;
; The valid values are:
;
; caller - Consider the callers preferred order ahead of the host's.
; host - Consider the host's preferred order ahead of the caller's.
; disabled - Disable the consideration of codec preference alltogether.
; (this is the original behaviour before preferences were added)
; reqonly - Same as disabled, only do not consider capabilities if
; the requested format is not available the call will only
; be accepted if the requested format is available.
;
; The default value is 'host'
;
;codecpriority=host
;
; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis.
;
;rtcachefriends=yes
;
; do not send the update request over realtime.
;
;rtnoupdate=yes
;
; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered when the registration expires
; the friend will vanish from the configuration until requested
; again. If set to an integer, friends expire
; within this number of seconds instead of the
; same as the registration interval
;
;rtautoclear=yes
;
;rtignoreexpire=yes ; when reading a peer from Realtime, if the peer's registration
; has expired based on its registration interval, used the stored
; address information regardless
; Guest sections for unauthenticated connection attempts. Just
; specify an empty secret, or provide no secret section.
;
[guest]
type=user
context=default
callerid="Guest IAX User"
;
; Trust Caller*ID Coming from iaxtel.com
;
[iaxtel]
type=user
context=default
auth=rsa
inkeys=iaxtel
;
; Trust Caller*ID Coming from iax.fwdnet.net
;
[iaxfwd]
type=user
context=default
auth=rsa
inkeys=freeworlddialup
;
; Trust callerid delivered over DUNDi/e164
;
;
;[dundi]
;type=user
;dbsecret=dundi/secret
;context=dundi-e164-local
;
; Further user sections may be added, specifying a context and a
; secret used for connections with that given authentication name.
; Limited IP based access control is allowed by use of "allow" and
; "deny" keywords. Multiple rules are permitted. Multiple permitted
; contexts may be specified, in which case the first will be the default.
; You can also override caller*ID so that when you receive a call you
; set the Caller*ID to be what you want instead of trusting what
; the remote user provides
;
; There are three authentication methods that are supported: md5, plaintext,
; and rsa. The least secure is "plaintext", which sends passwords cleartext
; across the net. "md5" uses a challenge/response md5 sum arrangement, but
; still requires both ends have plain text access to the secret. "rsa" allows
; unidirectional secret knowledge through public/private keys. If "rsa"
; authentication is used, "inkeys" is a list of acceptable public keys on the
; local system that can be used to authenticate the remote peer, separated by
; the ":" character. "outkey" is a single, private key to use to authenticate
; to the other side. Public keys are named /var/lib/asterisk/keys/<name>.pub
; while private keys are named /var/lib/asterisk/keys/<name>.key. Private
; keys should always be 3DES encrypted.
;
;
;[markster]
;type=user
;context=default
;context=local
;auth=md5,plaintext,rsa
;secret=markpasswd
;setvar=foo=bar
;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too
;notransfer=yes ; Disable IAX native transfer
;jitterbuffer=yes ; Override global setting an enable jitter buffer
; ; for this user
;callerid="Mark Spencer" <(256) 428-6275>
;deny=0.0.0.0/0.0.0.0
;accountcode=markster0101
;permit=209.16.236.73/255.255.255.0
;language=en ; Use english as default language
;
; Peers may also be specified, with a secret and
; a remote hostname.
;
[demo]
type=peer
username=asterisk
secret=supersecret
host=216.207.245.47
;sendani=no
;host=asterisk.linux-support.net
;port=5036
;mask=255.255.255.255
;qualify=yes ; Make sure this peer is alive
;jitterbuffer=no ; Turn off jitter buffer for this peer
;
; Peers can remotely register as well, so that they can be
; mobile. Default IP's can also optionally be given but
; are not required. Caller*ID can be suggested to the other
; side as well if it is for example a phone instead of another
; PBX.
;
;[dynamichost]
;host=dynamic
;secret=mysecret
;mailbox=1234 ; Notify about mailbox 1234
;inkeys=key1:key2
;peercontext=local ; Default context to request for calls to peer
;defaultip=216.207.245.34
;callerid="Some Host" <(256) 428-6011>
;
;
;[biggateway]
;type=peer
;host=192.168.0.1
;context=*
;secret=myscret
;trunk=yes ; Use IAX2 trunking with this host
;timezone=America/New_York ; Set a timezone for the date/time IE
;
;
; Friends are a short cut for creating a user and
; a peer with the same values.
;
;[marko]
;type=friend
;host=dynamic
;regexten=1234
;secret=moofoo
;context=default
;permit=0.0.0.0/0.0.0.0
[REMOTE_SERVER]
type=peer
host=192.168.5.3
secret=333
context=oficinamailinatorgdlzi
;
; IAX2 Provisioning Information
;
; Contains provisioning information for templates
; and for specific service entries.
;
; Templates provide a group of settings from which provisioning takes
; place. A template may be based upon any template that has been
; specified before it. If the template that an entry is based on is not
; specified then it is presumed to be 'default' (unless it is the first
; of course).
;
; Templates which begin with 'si-' are used for provisioning
; units with specific service identifiers. For example the
; entry "si-000364000126" would be used when the device with the
; corresponding service identifier of "000364000126" attempts
; to register or make a call.
;
[default]
;
; The port number the device should use to bind to. The default
; is 4569
;
;port=4569
;
; server is our PRIMARY server for registration and placing calls
;
;server=192.168.69.3
;
; altserver is the BACKUP server for registration and placing calls
; in the event the primary server is unavailable.
;
;altserver=192.168.69.4
;
; port is the port number to use for IAX2 outbound. The
; connections to the server and altserver -- default is of course
; 4569.
;serverport=4569
;
; language is the preferred language for the device
;
;language=en
;
; codec is the requested codec. The iaxy supports ulaw and adpcm
;
codec=ulaw
;
; flags is a comma separated list of flags which the device should
; use and may contain any of the following keywords:
;
; "register" - Register with server
; "secure" - Do not accept calls / provisioning not originated by the server
; "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
; "debug" - Output extra debugging to port 9999
;
; Note that use can use += and -= to adjust parameters
;
flags=register,heartbeat
;
; tos is the requested type of service setting and may be one a number or
; 'lowdelay','throughput','reliability','mincost' or 'none'
;
tos=lowdelay
;
; Example iaxy provisioning
;
;[si-000364000126]
;user=iaxy
;pass=bitsy
;flags += debug
;[si-000364000127]
;user=iaxy2
;pass=bitsy2
;template=si-000364000126
;flags += debug
;
;[*]
;
; If specified, the '*' provisioning is used for all devices which do
; not have another provisioning entry within the file. If unspecified, no
; provisioning will take place for devices which have no entry. DO NOT
; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
;
;template=default
; indications.conf
; Configuration file for location specific tone indications
; used by the pbx_indications module.
;
; NOTE:
; When adding countries to this file, please keep them in alphabetical
; order according to the 2-character country codes!
;
; The [general] category is for certain global variables.
; All other categories are interpreted as location specific indications
;
;
[general]
country=us ; default location
; [example]
; description = string
; The full name of your country, in English
; alias = iso[,iso]*
; List of other countries 2-letter iso codes, which have the same
; tone indications.
; ringcadance = num[,num]*
; List of durations the physical bell rings.
; dial = tonelist
; Set of tones to be played when one picks up the hook.
; busy = tonelist
; Set of tones played when the receiving end is busy.
; congestion = tonelist
; Set of tones played when there is some congestion (on the network?)
; callwaiting = tonelist
; Set of tones played when there is a call waiting in the background.
; dialrecall = tonelist
; Not well defined, many phone systems play a recall dial tone after hook flash
; record = tonelist
; Set of tones played when call recording is in progress
; info = tonelist
; Set of tones played with special information messages (e.g., "number is out of service")
; 'name' = tonelist
; Every other variable will be available as a shortcut for the "PlayList" command
; but will not automaticly be used by Asterisk.
;
;
; The tonelist itself is defined by a comma-separated sequence of elements.
; Each element consist of a frequency (f) with an optional duration (in ms)
; attached to it (f/duration). The frequency component may be a mixture of two
; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
; The implicit modulation depth is fixed at 90%, though.
; If the list element starts with a !, that element is NOT repeated,
; therefore, only if all elements start with !, the tonelist is time-limited,
; all others will repeat indefinitely.
;
; concisely:
; element = [!]freq[+|*freq2][/duration]
; tonelist = element[,element]*
;
; Please note that SPACES ARE NOT ALLOWED in tone lists!
;
[at]
description = Austria
ringcadance = 1000,5000
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
dial = 420
busy = 420/400,0/400
ring = 420/1000,0/5000
congestion = 420/200,0/200
callwaiting = 420/40,0/1960
dialrecall = 420
; RECORDTONE - not specified
record = 1400/80,0/14920
info = 950/330,1450/330,1850/330,0/1000
stutter = 380+420
[au]
description = Australia
; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf
; Normal Ring
ringcadance = 400,200,400,2000
; Distinctive Ring 1 - Forwarded Calls
; 400,400,200,200,400,1400
; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall
; 400,400,200,2000
; Distinctive Ring 3 - Multiple Subscriber Number 1
; 200,200,400,2200
; Distinctive Ring 4 - Selective Ring 1 + Centrex
; 400,2600
; Distinctive Ring 5 - Selective Ring 3
; 400,400,200,400,200,1400
; Distinctive Ring 6 - Multiple Subscriber Number 2
; 200,400,200,200,400,1600
; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy
; 200,400,200,400,200,1600
; Tones
dial = 413+438
busy = 425/375,0/375
ring = 413+438/400,0/200,413+438/400,0/2000
; XXX Congestion: Should reduce by 10 db every other cadence XXX
congestion = 425/375,0/375,420/375,0/375
callwaiting = 425/200,0/200,425/200,0/4400
dialrecall = 413+438
; Record tone used for Call Intrusion/Recording or Conference
record = !425/1000,!0/15000,425/360,0/15000
info = 425/2500,0/500
; Other Australian Tones
; The STD "pips" indicate the call is not an untimed local call
std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100
; Facility confirmation tone (eg. Call Forward Activated)
facility = 425
; Message Waiting "stutter" dialtone
stutter = 413+438/100,0/40
; Ringtone for calls to Telstra mobiles
ringmobile = 400+450/400,0/200,400+450/400,0/2000
[br]
description = Brazil
ringcadance = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion = 425/250,0/250,425/750,0/250
callwaiting = 425/50,0/1000
; Dialrecall not used in Brazil standard (using UK standard)
dialrecall = 350+440
; Record tone is not used in Brazil, use busy tone
record = 425/250,0/250
; Info not used in Brazil standard (using UK standard)
info = 950/330,1400/330,1800/330
stutter = 350+440
[be]
description = Belgium
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,3000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/3000
congestion = 425/167,0/167
callwaiting = 1400/175,0/175,1400/175,0/3500
; DIALRECALL - not specified
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440"
; RECORDTONE - not specified
record = 1400/500,0/15000
info = 900/330,1400/330,1800/330,0/1000
stutter = 425/1000,0/250
[ch]
description = Switzerland
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/200,425/200,0/4000
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; RECORDTONE - not specified
record = 1400/80,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425+340/1100,0/1100
[cl]
description = Chile
; According to specs from Telefonica CTC Chile
ringcadance = 1000,3000
dial = 400
busy = 400/500,0/500
ring = 400/1000,0/3000
congestion = 400/200,0/200
callwaiting = 400/250,0/8750
dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
record = 1400/500,0/15000
info = 950/333,1400/333,1800/333,0/1000
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
[cn]
description = China
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
dial = 450
busy = 450/350,0/350
ring = 450/1000,0/4000
congestion = 450/700,0/700
callwaiting = 450/400,0/4000
dialrecall = 450
record = 950/400,0/10000
info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
; STUTTER - not specified
stutter = 450+425
[cz]
description = Czech Republic
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
dial = 425/330,0/330,425/660,0/660
busy = 425/330,0/330
ring = 425/1000,0/4000
congestion = 425/165,0/165
callwaiting = 425/330,0/9000
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660
; RECORDTONE - not specified
record = 1400/500,0/14000
info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
; STUTTER - not specified
stutter = 425/450,0/50
[de]
description = Germany
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
dial = 425
busy = 425/480,0/480
ring = 425/1000,0/4000
congestion = 425/240,0/240
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
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; RECORDTONE - not specified
record = 1400/80,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425+400
[dk]
description = Denmark
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; RECORDTONE - not specified
record = 1400/80,0/15000
info = 950/330,1400/330,1800/330,0/1000
; STUTTER - not specified
stutter = 425/450,0/50
[ee]
description = Estonia
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
dial = 425
busy = 425/300,0/300
ring = 425/1000,0/4000
congestion = 425/200,0/200
; CALLWAIT not in accordance to ITU
callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
; DIALRECALL - not specified
dialrecall = 425/650,0/25
; RECORDTONE - not specified
record = 1400/500,0/15000
; INFO not in accordance to ITU
info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
; STUTTER not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[fi]
description = Finland
ringcadance = 1000,4000
dial = 425
busy = 425/300,0/300
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/150,0/150,425/150,0/8000
dialrecall = 425/650,0/25
record = 1400/500,0/15000
info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
stutter = 425/650,0/25
[fr]
description = France
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1500,3500
; Dialtone can also be 440+330
dial = 440
busy = 440/500,0/500
ring = 440/1500,0/3500
; CONGESTION - not specified
congestion = 440/250,0/250
callwait = 440/300,0/10000
; DIALRECALL - not specified
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
; RECORDTONE - not specified
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330
stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440
[gr]
description = Greece
ringcadance = 1000,4000
dial = 425/200,0/300,425/700,0/800
busy = 425/300,0/300
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/150,0/150,425/150,0/8000
dialrecall = 425/650,0/25
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 425/650,0/25
[hu]
description = Hungary
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1250,3750
dial = 425
busy = 425/300,0/300
ring = 425/1250,0/3750
congestion = 425/300,0/300
callwaiting = 425/40,0/1960
dialrecall = 425+450
; RECORDTONE - not specified
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 350+375+400
[it]
description = Italy
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 425/200,0/200,425/600,0/1000
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 470/400,425/400
[lt]
description = Lithuania
ringcadence = 1000,4000
dial = 425
busy = 425/350,0/350
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/150,0/150,425/150,0/4000
; DIALRECALL - not specified
dialrecall = 425/500,0/50
; RECORDTONE - not specified
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
; STUTTER - not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[mx]
description = Mexico
ringcadance = 2000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/200,0/600,425/200,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
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
[nl]
description = Netherlands
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadance = 1000,4000
; Most of these 425's can also be 450's
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/500,0/9500
; DIALRECALL - not specified
dialrecall = 425/500,0/50
; RECORDTONE - not specified
record = 1400/500,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425/500,0/50
[no]
description = Norway
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/600,425/200,0/10000
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 470/400,425/400
[nz]
description = New Zealand
;NOTE - the ITU has different tonesets for NZ, but according to some residents there,
; this is, indeed, the correct way to do it.
ringcadence = 400,200,400,2000
dial = 400
busy = 400/250,0/250
ring = 400+450/400,0/200,400+450/400,0/2000
congestion = 400/375,0/375
callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
dialrecall = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,400
record = 1400/425,0/15000
info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
stutter = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,!400/100!0/100,!400/100,!0/100,!400/100,!0/100,400
[pl]
description = Poland
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/500,0/500
callwaiting = 425/150,0/150,425/150,0/4000
; DIALRECALL - not specified
dialrecall = 425/500,0/50
; RECORDTONE - not specified
record = 1400/500,0/15000
; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000
; STUTTER - not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[pt]
description = Portugal
ringcadance = 1000,5000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/5000
congestion = 425/200,0/200
callwaiting = 440/300,0/10000
dialrecall = 425/1000,0/200
record = 1400/500,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[ru]
description = Russia / ex Soviet Union
ringcadance = 800,3200
dial = 425
busy = 425/350,0/350
ring = 425/800,0/3200
congestion = 425/350,0/350
callwaiting = 425/200,0/5000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
[se]
description = Sweden
ringcadance = 1000,5000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425/250,0/750
callwaiting = 425/200,0/500,425/200,0/9100
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
record = 1400/500,0/15000
info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; stutter = 425/320,0/20 ; Real swedish standard, not used for now
[sg]
description = Singapore
; Singapore
; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz
ringcadence = 400,200,400,2000
dial = 425
ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90%
busy = 425/750,0/750
congestion = 425/250,0/250
callwaiting = 425*24/300,0/200,425*24/300,0/3200
stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
info = 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference
dialrecall = 425*24/500,0/500,425/500,0/2500 ; unspecified in IDA reference, use repeating Holding Tone A,B
record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s
; additionally defined in reference
nutone = 425/2500,0/500
intrusion = 425/250,0/2000
warning = 425/624,0/4376 ; end of period tone, warning
acceptance = 425/125,0/125
holdinga = !425*24/500,!0/500 ; followed by holdingb
holdingb = !425/500,!0/2500
[uk]
description = United Kingdom
ringcadance = 400,200,400,2000
; These are the official tones taken from BT SIN350. The actual tones
; used by BT include some volume differences so sound slightly different
; from Asterisk-generated ones.
dial = 350+440
; Special dial is the intermittent dial tone heard when, for example,
; you have a divert active on the line
specialdial = 350+440/750,440/750
; Busy is also called "Engaged"
busy = 400/375,0/375
; "Congestion" is the Beep-bip engaged tone
congestion = 400/400,0/350,400/225,0/525
; "Special Congestion" is not used by BT very often if at all
specialcongestion = 400/200,1004/300
unobtainable = 400
ring = 400+450/400,0/200,400+450/400,0/2000
callwaiting = 400/100,0/4000
; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it
; is and I've not used a payphone for years)
creditexpired = 400/125,0/125
; These two are used to confirm/reject service requests on exchanges that
; don't do voice announcements.
confirm = 1400
switching = 400/200,0/400,400/2000,0/400
; This is the three rising tones Doo-dah-dee "Special Information Tone",
; usually followed by the BT woman saying an appropriate message.
info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
; Not listed in SIN350
record = 1400/500,0/60000
stutter = 350+440/750,440/750
[us]
description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
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
[us-old]
description = United States Circa 1950/ North America
ringcadance = 2000,4000
dial = 600*120
busy = 500*100/500,0/500
ring = 420*40/2000,0/4000
congestion = 500*100/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
[tw]
description = Taiwan
; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
; http://www.pbx-manufacturer.com/ky120dx.htm
; http://www.nettwerked.net/tones.txt
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
;
; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
; ringing tone 440+480Hz 1 sec. on ,2 sec. off
;
ringcadance = 1000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/1000,0/2000
congestion = 480+620/250,0/250
callwaiting = 350+440/250,0/250,350+440/250,0/3250
dialrecall = 300/1500,0/500
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
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
[za]
description = South Africa
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
; (definitions for other countries can also be found there)
; Note, though, that South Africa uses two switch types in their network - Alcatel
; switches - mainly in the Western Cape, and Siemens elsewhere.
; The former use 383+417 in dial, ringback etc. The latter use 400*33
; I've provided both, uncomment the ones you prefer
ringcadance = 400,200,400,2000
; dial/ring/callwaiting for the Siemens switches:
dial = 400*33
ring = 400*33/400,0/200,400*33/400,0/2000
callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
; dial/ring/callwaiting for the Alcatel switches:
; dial = 383+417
; ring = 383+417/400,0/200,383+417/400,0/2000
; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
congestion = 400/250,0/250
busy = 400/500,0/500
dialrecall = 350+440
; XXX Not sure about the RECORDTONE
record = 1400/500,0/10000
info = 950/330,1400/330,1800/330,0/330
stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33
[general]
;debug=yes ;;Turn on debugging by default.
;autoprune=yes ;;Auto remove users from buddy list.
;autoregister=yes ;;Auto register users from buddy list.
;[asterisk] ;;label
;type=client ;;Client or Component connection
;serverhost=astjab.org ;;Route to server for example,
;; talk.google.com
;username=asterisk@astjab.org/asterisk ;;Username with optional roster.
;secret=blah ;;Password
;port=5222 ;;Port to use defaults to 5222
;usetls=yes ;;Use tls or not
;usesasl=yes ;;Use sasl or not
;buddy=mogorman@astjab.org ;;Manual addition of buddy to list.
;statusmessage="I am available" ;;Have custom status message for
;;Asterisk.
;timeout=100 ;;Timeout on the message stack.
;
; Logging Configuration
;
; In this file, you configure logging to files or to
; the syslog system.
;
; "logger reload" at the CLI will reload configuration
; of the logging system.
[general]
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
; see strftime(3) Linux manual for format specifiers
;dateformat=%F %T
;
; This appends the hostname to the name of the log files.
;appendhostname = yes
;
; This determines whether or not we log queue events to a file (defaults to yes).
;queue_log = no
;
; This determines whether or not we log generic events to a file (defaults to yes).
;event_log = no
;
;
; For each file, specify what to log.
;
; For console logging, you set options at start of
; Asterisk with -v for verbose and -d for debug
; See 'asterisk -h' for more information.
;
; Directory for log files is configures in asterisk.conf
; option astlogdir
;
[logfiles]
;
; Format is "filename" and then "levels" of debugging to be included:
; debug
; notice
; warning
; error
; verbose
;
; Special filename "console" represents the system console
;
;debug => debug
console => notice,warning,error,debug,verbose
;console => notice,warning,error,debug
messages => notice,warning,error,debug,verbose
;full => notice,warning,error,debug,verbose
;syslog keyword : This special keyword logs to syslog facility
;
;syslog.local0 => notice,warning,error
;
;
; AMI - The Asterisk Manager Interface
;
; Third party application call management support
; and PBX event supervision
;
; This configuration file is read every time someone
; logs in
;
; Use the "show manager commands" at the CLI to list
; availabale manager commands and their authorization
; levels.
;
; "show manager command <command>" will show a help text.
;
; ------------------- SECURITY NOTE -----------------
; Note that you should not enable the AMI on a public
; IP address. If needed, block this TCP port with
; iptables (or another FW software) and reach it
; with IPsec, SSH or SSL vpn tunnel
;
[general]
enabled = no
port = 5038
bindaddr = 0.0.0.0
;displayconnects = yes
;[mark]
;secret = mysecret
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.73/255.255.255.0
;
; Authorization for various classes
;read = system,call,log,verbose,command,agent,user
;write = system,call,log,verbose,command,agent,user
;
; Configuration file for MeetMe simple conference rooms
; for Asterisk of course.
;
; This configuration file is read every time you call app meetme()
;
[rooms]
;
; Usage is conf => confno[,pin][,adminpin]
;
;conf => 1234
;conf => 2345,9938
;
; MGCP Configuration for Asterisk
;
[general]
;port = 2427
;bindaddr = 0.0.0.0
;[dlinkgw]
;host = 192.168.0.64
;context = default
;canreinvite = no
;line => aaln/2
;line => aaln/1
;; The MGCP channel supports the following service codes:
;; # - Transfer
;; *67 - Calling Number Delivery Blocking
;; *70 - Cancel Call Waiting
;; *72 - Call Forwarding Activation
;; *73 - Call Forwarding Deactivation
;; *78 - Do Not Disturb Activation
;; *79 - Do Not Disturb Deactivation
;; *8 - Call pick-up
;
; known to work with Swissvoice IP10s
;[192.168.1.20]
;context=local
;host=192.168.1.20
;callerid = "John Doe" <123>
;callgroup=0
;pickupgroup=0
;nat=no
;threewaycalling=yes
;transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
;callwaiting=yes ; this might be a cause of trouble for ip10s
;cancallforward=yes
;line => aaln/1
;
;[dph100]
;
; Supporting the DPH100M requires defining DLINK_BUGGY_FIRMWARE in
; chan_mgcp.c in addition to enabling the slowsequence mode due to
; bugs in the D-Link firmware
;
;context=local
;host=dynamic
;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or 'hybrid' which starts in none and moves to inband. Default is none.
;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing
;line => aaln/1
; known to work with wave7optics FTTH LMGs
;[192.168.1.20]
;accountcode = 1000 ; record this in cdr as account identification for billing
;amaflags = billing ; record this in cdr as flagged for 'billing', 'documentation', or 'omit'
;context = local
;host = 192.168.1.20
;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*' another common format is '*'
;callerid = "Duane Cox" <123> ; now lets setup line 1 using per endpoint configuration...
;callwaiting = no
;callreturn = yes
;cancallforward = yes
;canreinvite = no
;transfer = no
;dtmfmode = inband
;line => aaln/1 ; now lets save this config to line1 aka aaln/1
;callerid = "Duane Cox" <456> ; now lets setup line 2
;callwaiting = no
;callreturn = yes
;cancallforward = yes
;canreinvite = no
;transfer = no
;dtmfmode = inband
;line => aaln/2 ; now lets save this config to line2 aka aaln/2
;
; chan_misdn sample config
;
; general section:
;
; for debugging and general setup, things that are not bound to port groups
;
[general]
;
; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
;
misdn_init=/etc/misdn-init.conf
; set debugging flag:
; 0 - No Debug
; 1 - mISDN Messages and * - Messages, and * - State changes
; 2 - Messages + Message specific Informations (e.g. bearer capability)
; 3 - very Verbose, the above + lots of Driver specific infos
; 4 - even more Verbose than 3
;
; default value: 0
;
debug=0
; set debugging file and flags for mISDNuser (NT-Stack)
;
; flags can be or'ed with the following values:
;
; DBGM_NET 0x00000001
; DBGM_MSG 0x00000002
; DBGM_FSM 0x00000004
; DBGM_TEI 0x00000010
; DBGM_L2 0x00000020
; DBGM_L3 0x00000040
; DBGM_L3DATA 0x00000080
; DBGM_BC 0x00000100
; DBGM_TONE 0x00000200
; DBGM_BCDATA 0x00000400
; DBGM_MAN 0x00001000
; DBGM_APPL 0x00002000
; DBGM_ISDN 0x00004000
; DBGM_SOCK 0x00010000
; DBGM_CONN 0x00020000
; DBGM_CDATA 0x00040000
; DBGM_DDATA 0x00080000
; DBGM_SOUND 0x00100000
; DBGM_SDATA 0x00200000
; DBGM_TOPLEVEL 0x40000000
; DBGM_ALL 0xffffffff
;
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
; the big trace
;
; default value: [not set]
;
;tracefile=/var/log/asterisk/misdn.log
; set to yes if you want mISDN_dsp to bridge the calls in HW
;
; default value: yes
;
bridging=no
;
; watches the L1s of every port. If one l1 is down it tries to
; get it up. The timeout is given in seconds. with 0 as value it
; does not watch the l1 at all
;
; default value: 0
;
; this option is only read at loading time of chan_misdn,
; which means you need to unload and load chan_misdn to change the
; value, an asterisk restart should do the trick
;
l1watcher_timeout=0
; stops dialtone after getting first digit on nt Port
;
; default value: yes
;
stop_tone_after_first_digit=yes
; whether to append overlapdialed Digits to Extension or not
;
; default value: yes
;
append_digits2exten=yes
;;; CRYPTION STUFF
; Whether to look for dynamic crypting attempt
;
; default value: no
;
dynamic_crypt=no
; crypt_prefix, what is used for crypting Protocol
;
; default value: [not set]
;
crypt_prefix=**
; Keys for cryption, you reference them in the dialplan
; later also in dynamic encr.
;
; default value: [not set]
;
crypt_keys=test,muh
; users sections:
;
; name your sections as you which but not "general" !
; the sections are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name,
; chan_misdn tries every port in this section to find a
; new free channel
;
; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;
[default]
; define your default context here
;
; default value: default
;
context=misdn
; language
;
; default value: en
;
language=en
;
; sets the musiconhold class
;
musicclass=default
;
; Either if we should produce DTMF Tones ourselves
;
senddtmf=yes
;
; If we should generate Ringing for chan_sip and others
;
far_alerting=no
;
; here you can define which bearers should be allowed
;
allowed_bearers=all
; Prefixes for national and international, those are put before the
; oad if an according dialplan is set by the other end.
;
; default values: nationalprefix : 0
; internationalprefix : 00
;
nationalprefix=0
internationalprefix=00
; set rx/tx gains between -8 and 8 to change the RX/TX Gain
;
; default values: rxgain: 0
; txgain: 0
;
rxgain=0
txgain=0
; some telcos especially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
;
te_choose_channel=no
;
; This option defines, if chan_misdn should check the L1 on a PMP
; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
; as well, since chan_misdn has no chance to distinguish if the L1 is down
; because of a lost Link or because the Provider shut it down...
;
; default: no
;
pmp_l1_check=no
pp_l2_check=no
;
; in PMP this option defines which cause should be sent out to
; the 3. caller. chan_misdn does not support callwaiting on TE
; PMP side. This allows to modify the RELEASE_COMPLETE cause
; at least.
;
reject_cause=16
;
; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
; this requests additional Infos, so we can waitfordigits
; without much issues. This works only for PTP Ports
;
; default value: no
;
need_more_infos=no
;
; set this to yes if you want to disconnect calls when a timeout occurs
; for example during the overlapdial phase
;
nttimeout=no
; set the method to use for channel selection:
; standard - always choose the first free channel with the lowest number
; round_robin - use the round robin algorithm to select a channel. use this
; if you want to balance your load.
;
; default value: standard
;
method=standard
;
; dialplan means Type Of Number in ISDN Terms (for outgoing calls)
;
; there are different types of the dialplan:
;
; dialplan -> outgoing Number
; localdialplan -> callerid
; cpndialplan -> connected party number
;
; dialplan options:
;
; 0 - unknown
; 1 - International
; 2 - National
; 4 - Subscriber
;
; This setting is used for outgoing calls
;
; default value: 0
;
dialplan=0
localdialplan=0
cpndialplan=0
;
; turn this to no if you don't mind correct handling of Progress Indicators
;
early_bconnect=yes
;
; turn this on if you like to send Tone Indications to a Incoming
; isdn channel on a TE Port. Rarely used, only if the Telco allows
; you to send indications by yourself, normally the Telco sends the
; indications to the remote party.
;
; default: no
;
incoming_early_audio=no
; uncomment the following to get into s extension at extension conf
; there you can use DigitTimeout if you can't or don't want to use
; isdn overlap dial.
; note: This will jump into the s exten for every exten!
;
; default value: no
;
;always_immediate=no
;
; set this to yes if you want to generate your own dialtone
; with always_immediate=yes, else chan_misdn generates the dialtone
;
; default value: no
;
nodialtone=no
; uncomment the following if you want callers which called exactly the
; base number (so no extension is set) jump to the s extension.
; if the user dials something more it jumps to the correct extension
; instead
;
; default value: no
;
;immediate=no
; uncomment the following to have hold and retrieve support
;
; default value: no
;
;hold_allowed=yes
; Pickup and Callgroup
;
; default values: not set = 0
; range: 0-63
;
;callgroup=1
;pickupgroup=1
;
; these are the exact isdn screening and presentation indicators
; if -1 is given for both values the presentation indicators are used
; from asterisks SetCallerPres application.
; s=0, p=0 -> callerid presented not screened
; s=1, p=1 -> callerid presented but screened (the remote end does not see it!)
;
; default values s=-1, p=-1
presentation=-1
screen=-1
; this enables echocancellation, with the given number of taps
; be aware, move this setting only to outgoing portgroups!
; A value of zero turns echocancellation off.
;
; possible values are: 0,32,64,128,256,yes(=128),no(=0)
;
; default value: no
;
;echocancel=no
; this disables echocancellation when the call is bridged between
; mISDN channels
;
; default value: no
;
echocancelwhenbridged=no
; Set this to no to disable echotraining. You can enter a number > 10
; the value is a multiple of 0.125 ms.
;
; default value: no
; yes = 2000
; no = 0
;
echotraining=no
;
; chan_misdns jitterbuffer, default 4000
;
jitterbuffer=4000
;
; change this threshold to enable dejitter functionality
;
jitterbuffer_upper_threshold=0
;
; change this to yes, if you want to bridge a mISDN data channel to
; another channel type or to an application.
;
hdlc=no
;
; defines the maximum amount of incoming calls per port for
; this group. Calls which exceed the maximum will be marked with
; the channel variable MAX_OVERFLOW. It will contain the amount of
; overflowed calls
;
max_incoming=-1
;
; defines the maximum amount of outgoing calls per port for this group
; exceeding calls will be rejected
;
max_outgoing=-1
[intern]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
ports=1,2
; context where to go to when incoming Call on one of the above ports
context=Intern
[internPP]
;
; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
; configs. For backwards compatibility you can still set ptp here.
;
ports=3
[first_extern]
; again port defs
ports=4
; again a context for incoming calls
context=Extern1
; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
; here you can give a comma separated list or simply an '*' for
; any msn.
msns=*
; here an example with given msns
[second_extern]
ports=5
context=Extern2
callerid=15
msns=102,144,101,104
;
; isdn4linux
;
; Configuration file
;
[interfaces]
;
; By default, incoming calls should come in on the "remote" context
;
context=remote
;
; Modem Drivers to load
;
driver=aopen ; modem by AOpen
;driver=i4l ; isdn4linux - an alternative to i4l is to use chan_capi
;
; Default language
;
;language=en
;
; We can optionally override the auto detection. This is necessary
; particularly if asterisk does not know about our kind of modem.
;
;type=autodetect
;type=aopen
;
; We can strip a given number of digits on outgoing dialing, so, for example
; you can have it dial "8871042" when given "98871042".
;
stripmsd=0
;
; Type of dialing
;
dialtype=tone
;dialtype=pulse
;
; Mode selection. "Immediate" means that as soon as you dial, you're connected
; and the line is considered up. "Ring" means we wait until the ring cadence
; occurs at least once. "Answer" means we wait until the other end picks up.
;
;mode=answer
;mode=ring
mode=immediate
;
; List all devices we can use.
;
;device => /dev/ttyS3
;
; ISDN example (using i4l)
;
;msn=39907835
;device => /dev/ttyI0
;===============
; More complex ISDN example
;
; A single device which listens to 3 MSNs
; the wildcard '*' can be used when all MSN's should be accepted.
; (The incoming number can be used to go directly into the extension
; with the matching number. I.e. if MSN 33 is called, (context,33)
; will tried first, than (context,s) and finally (default,s).
;
;msn=50780020
;incomingmsn=50780020,50780021,50780022
;device => /dev/ttyI2
;
; If set, only these numbers are allowed to be set as A number
; when making an outbound call. callerid is used to set A
; number.
;outgoingmsn=50780023,50780024
;
; Set DTMF-detection/generation mode to:
; asterisk: Let Asterisk do inband detection (default)
; i4l: Use the inband detection made by ISDN4Linux
; none: Don't detect inband DTMF
; both: Transmit using both in-band and out of band (generation only)
;
; You may specify either one mode, or the detection/generation mode
; individually separated by a '/'.
;
;dtmfmode=asterisk ; Detect using Asterisk
;dtmfmode=asterisk/both ; Detect using Asterisk, generate w/ both
; two other devices, which are in group '1' and are used when an
; outgoing dial used: exten => s,1,Dial,Modem/g1:1234|60|r
; (we do not need more outgoing devices, since ISDN2 has only 2 channels.)
; Lines can be in more than one group (0-63); comma separated list.
;
group=1 ; group=1,2,3,9-12
;msn=50780023
;device => /dev/ttyI3
;device => /dev/ttyI4
;
; 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
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
load => chan_modem.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
chan_modem.so=yes
;
; Music on hold class definitions
;
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z
;unbuffered => mp3nb:/var/lib/asterisk/mohmp3
;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3
; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces)
;manual => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
;
; File-based (native) music on hold
;
; This plays files directly from the specified directory, no external
; processes are required. Files are played in normal sorting order
; (same as a sorted directory listing), and no volume or other
; sound adjustments are available. If the file is available in
; the same format as the channel's codec, then it will be played
; without transcoding (same as Playback would do in the dialplan).
; Files can be present in as many formats as you wish, and the
; 'best' format will be chosen at playback time.
;
; NOTE:
; If you are not using "autoload" in modules.conf, then you
; must ensure that the format modules for any formats you wish
; to use are loaded _before_ res_musiconhold. If you do not do
; this, res_musiconhold will skip the files it is not able to
; understand when it loads.
;
[moh_files]
;native => /var/lib/asterisk/moh-native
;native-random => /var/lib/asterisk/moh-native,r
#
# Sample muted configuration file
#
# Copyright (C) 2004 Digium, Inc.
#
# What is this? Well, haven't you ever wished you could automatically
# turn down the volume on your stereo, CDPlayer, etc, when a call comes in,
# and then return it to normal when the call ends? Well, this is a possible
# mechanism to make this happen!
# You have to fire up the new utils/muted, which runs as a daemon in the
# background. This daemon connects to asterisk via a manager interface, and
# also reads this config file. When the channels mentioned
# are activated, it tweaks the sound levels on the sound card(s).
# So, depending on the sound card, you may be able to run all your sound
# generating devices thru your sound card, and use this mechanism to quiet
# them down while you are on the phone. If anyone figures out how to make
# this work with kids, please inform!!
#
# First you have the host, username, and password
# we use to connect to the asterisk system
#
host localhost
user user
pass pass
#
# List each channel we're supposed to watch
#
channel Zap/1
channel Zap/2
channel SIP/mark
#
# Mute level is the percentage of the current volume we should
# lower the music to.
#
mutelevel 20
#
# Smooth fade makes the fadein/fadeout nicer sounding
#
smoothfade
;
; Open Settlement Protocol Sample Configuration File
;
;
; This file contains configuration of providers that
; are used by the OSP subsystem of Asterisk. The section
; "general" is reserved for global options. Each other
; section declares an OSP Provider. The provider "default"
; is used when no provider is otherwise specified.
;
[general]
;
; Should hardware accelleration be enabled? May not be changed
; on a reload.
;
;accelerate=yes
;[default]
;
; All paths are presumed to be under /var/lib/asterisk/keys unless
; the path begins with '/'
;
; Specify the private keyfile. If unspecified, defaults to the name
; of the section followed by "-privatekey.pem" (e.g. default-privatekey.pem)
;
;privatekey=pkey.pem
;
; Specify the local certificate file. If unspecified, defaults to
; the name of the section followed by "-localcert.pem"
;
;localcert=localcert.pem
;
; Specify one or more Certificate Authority keys. If none are listed,
; a single one is added with the name "-cacert.pem"
;
;cacert=cacert_0.pem
;
; Specific parameters can be tuned as well:
;
; maxconnections: Max number of simultaneous connections to the provider (default=20)
; retrydelay: Extra delay between retries (default=0)
; retrylimit: Max number of retries before giving up (default=2)
; timeout: Timeout for response in milliseconds (default=500)
;
;maxconnections=20
;retrydelay=0
;retrylimit=2
;timeout=500
;
; List all service points for this provider
;
;servicepoint=http://osptestserver.transnexus.com:1080/osp
;
; Set the "source" for requesting authorization
;
;source=foo
;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console? Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=local
;
; Default extension to call
;
extension=s
;
; Default language
;
;language=en
;
; Silence supression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary. Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth between
; read and write modes. Unfortunately, this fails sometimes on older hardware.
; To prevent the driver from switching (ie. only play files on your speakers),
; then set the playbackonly option to yes. Default is no. Note this option has
; no effect on full-duplex cards.
;playbackonly=no
;
; Linux Telephony Interface
;
; Configuration file
;
[interfaces]
;
; Select a mode, either the phone jack provides dialtone, reads digits,
; then starts PBX with the given extension (dialtone mode), or
; immediately provides the PBX without reading any digits or providing
; any dialtone (this is the immediate mode, the default). Also, you
; can set the mode to "fxo" if you have a linejack to make it operate
; properly.
;
mode=immediate
;mode=dialtone
;mode=fxo
;
; You can decide which format to use by default, "g723.1" or "slinear".
; XXX Be careful, sometimes the card causes kernel panics when running
; in signed linear mode for some reason... XXX
;
format=slinear
;format=g723.1
;
; And set the echo cancellation to "off", "low", "medium", and "high".
; This is not supported on all phones.
;
echocancel=medium
;
; You can optionally use VAD/CNG silence supression
;
;silencesupression=yes
;
; List all devices we can use. Contexts may also be specified
;
;context=local
;
; You can set txgain and rxgain for each device in the same way as context.
; If you want to change default gain value (1.0 =~ 100%) for device, simple
; add txgain or rxgain line before device line. But rememeber, if you change
; volume all cards listed below will be affected by these values. You can
; use float values (1.0, 0.5, 2.0) or percentage values (100%, 150%, 50%).
;
;txgain=100%
;rxgain=1.0
;device => /dev/phone0
[general]
maxretries = 2 ;How many chances the caller has to enter their number
[general]
;
; Global settings for call queues
;
; Persistent Members
; Store each dynamic agent in each queue in the astdb so that
; when asterisk is restarted, each agent will be automatically
; readded into their recorded queues. Default is 'yes'.
;
persistentmembers = yes
;
; Note that a timeout to fail out of a queue may be passed as part of application call
; from extensions.conf:
; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout])
; example: Queue(dave|t|||45)
;[markq]
;
; A sample call queue
;
; Musiconhold sets which music applies for this particular
; call queue (configure classes in musiconhold.conf)
;
;musiconhold = default
;
; An announcement may be specified which is played for the member as
; soon as they answer a call, typically to indicate to them which queue
; this call should be answered as, so that agents or members who are
; listening to more than one queue can differentiated how they should
; engage the customer
;
;announce = queue-markq
;
; A strategy may be specified. Valid strategies include:
;
; ringall - ring all available channels until one answers (default)
; roundrobin - take turns ringing each available interface
; leastrecent - ring interface which was least recently called by this queue
; fewestcalls - ring the one with fewest completed calls from this queue
; random - ring random interface
; rrmemory - round robin with memory, remember where we left off last ring pass
;
;strategy = ringall
;
; Second settings for service level (default 0)
; Used for service level statistics (calls completed within service level time frame)
;servicelevel = 60
;
; A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
;context = qoutcon
;
; How long do we let the phone ring before we consider this a timeout...
;
;timeout = 15
;
; How long do we wait before trying all the members again?
;
;retry = 5
;
; Weight of queue - when compared to other queues, higher weights get
; first shot at available channels when the same channel is included in
; more than one queue.
;
;weight=0
;
; After a successful call, how long to wait before sending a potentially
; free member another call (default is 0, or no delay)
;
;wrapuptime=15
;
; Maximum number of people waiting in the queue (0 for unlimited)
;
;maxlen = 0
;
;
; How often to announce queue position and/or estimated holdtime to caller (0=off)
;
;announce-frequency = 90
;
; Should we include estimated hold time in position announcements?
; Either yes, no, or only once.
; Hold time will be announced as the estimated time,
; or "less than 2 minutes" when appropriate.
;
;announce-holdtime = yes|no|once
;
; What's the rounding time for the seconds?
; If this is non zero then we announce the seconds as well as the minutes rounded to this value
;
; announce-round-seconds = 10
;
; Use these sound files in making position/holdtime announcements. The
; defaults are as listed below -- change only if you need to.
;
;queue-youarenext = queue-youarenext ; ("You are now first in line.")
;queue-thereare = queue-thereare ; ("There are")
;queue-callswaiting = queue-callswaiting ; ("calls waiting.")
;queue-holdtime = queue-holdtime ; ("The current est. holdtime is")
;queue-minutes = queue-minutes ; ("minutes.")
;queue-seconds = queue-seconds ; ("seconds.")
;queue-thankyou = queue-thankyou ; ("Thank you for your patience.")
;queue-lessthan = queue-less-than ; ("less than")
;queue-reporthold = queue-reporthold ; ("Hold time")
;
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
; recorded, and you are not recording while people are listening to MOH.
; To enable monitoring, simply specify "monitor-format"; it will be disabled
; otherwise.
;
; You can specify the monitor filename with by calling Set(MONITOR_FILENAME=foo)
; Otherwise it will use ${UNIQUEID}
;
; monitor-format = gsm|wav|wav49
;
; If you wish to have the two files joined together when the call ends set this to yes
;
; monitor-join = yes
;
; This setting controls whether callers can join a queue with no members. There are three
; choices:
;
; yes - callers can join a queue with no members or only unavailable members
; no - callers cannot join a queue with no members
; strict - callers cannot join a queue with no members or only unavailable members
;
; joinempty = yes
;
; If you wish to remove callers from the queue when new callers cannot join, set this setting
; to one of the same choices for 'joinempty'
;
; leavewhenempty = yes
;
;
; If this is set to yes, the following manager events will be generated:
; AgentCalled, AgentDump, AgentConnect, AgentComplete
; (may generate some extra manager events, but probably ones you want)
;
; eventwhencalled = yes
;
; If this is set to no, the following manager events will be generated:
; QueueMemberStatus
; (may generate a WHOLE LOT of extra manager events)
;
; eventmemberstatusoff = no
;
; If you wish to report the caller's hold time to the member before they are connected
; to the caller, set this to yes.
;
; reportholdtime = no
;
;
; If you wish to have a delay before the member is connected to the caller (or before the member
; hears any announcement messages), set this to the number of seconds to delay.
;
; memberdelay = 0
;
; If timeoutrestart is set to yes, then the timeout for an agent to answer is
; reset if a BUSY or CONGESTION is received. This can be useful if agents
; are able to cancel a call with reject or similar.
;
; timeoutrestart = no
;
; Each member of this call queue is listed on a separate line in
; the form technology/dialstring. "member" means a normal member of a
; queue. An optional penalty may be specified after a comma, such that
; entries with higher penalties are considered last.
;
;member => Zap/1
;member => Zap/2
;member => Agent/1001
;member => Agent/1002
;
; Note that using agent groups is probably not what you want. Strategies do
; not propagate down to the Agent system so if you want round robin, least
; recent, etc, you should list all the agents in this file individually and not
; use agent groups.
;
;member => Agent/@1 ; Any agent in group 1
;member => Agent/:1,1 ; Any agent in group 1, wait for first
; available, but consider with penalty
;
; Sample configuration for res_config_mysql.c
;
; The value of dbhost may be either a hostname or an IP address.
; If dbhost is commented out or the string "localhost", a connection
; to the local host is assumed and dbsock is used instead of TCP/IP
; to connect to the server.
;
[general]
;dbhost = 127.0.0.1
;dbname = asterisk
;dbuser = myuser
;dbpass = mypass
;dbport = 3306
;dbsock = /tmp/mysql.sock
;;; odbc setup file
[asterisk]
dsn => asterisk
;username => myuser
;password => mypass
pre-connect => yes
;[mysql2]
;dsn => MySQL-asterisk
;username => myuser
;password => mypass
;pre-connect => yes
;
; Configuration file for res_snmp
;
[general]
; We run as a subagent per default -- to run as a full agent
; we must run as root (to be able to bind to port 161)
;subagent = yes
; SNMP must be explicitly enabled to be active
;enabled = yes
; Radio Repeater / Remote Base configuration file (for use with app_rpt)
;
;[000] ; Node ID of first repeater
;rxchannel = Zap/1 ; Rx audio/signalling channel
; Note: if you use a unified interface (tx/rx on one channel), only
; specify the rxchannel and the txchannel will be assumed from the rxchannel
;txchannel = Zap/2 ; Tx audio/signalling channel
;functions = functions-repeater ; DTMF function list
;; specify this for a different function list then local when on link
;;link_functions = functions-different ; DTMF function list for link
;tonezone = us ; use US tones (default)
;context = default ; dialing context for phone
;callerid = "WB6NIL Repeater" <(559) 692-1234> ; Callerid for phone calls
;idrecording = wb6nil ; id recording
;accountcode=RADIO ; account code (optional)
;funcchar = * ; function lead-in character (defaults to '*')
;endchar = # ; command mode end character (defaults to '#')
;hangtime=1000 ; squelch tail hang time (in ms) (optional)
;totime=100000 ; transmit time-out time (in ms) (optional)
;idtime=30000 ; id interval time (in ms) (optional)
;politeid=30000 ; time in milliseconds before ID timer
; expires to try and ID in the tail.
; (optional, default is 30000).
;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none
;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none
; The default values for hangtime, time-out time, and id interval time are
; 5 seconds (5000 ms), 3 minutes (180000 ms), and 5 minutes (300000 ms)
; respectively
;[001] ; Node ID of second repeater
;rxchannel = Zap/3 ; Rx audio/signalling channel
; Note: if you use a unified interface (tx/rx on one channel), only
; specify the rxchannel and the txchannel will be assumed from the rxchannel
;txchannel = Zap/4 ; Tx audio/signalling channel
;tonezone = us ; use US tones (default)
;context = default ; dialing context for phone
;callerid = "WB6NIL Repeater" <(559) 692-1234> ; Callerid for phone calls
;idrecording = wb6nil ; id recording
;accountcode=RADIO ; account code (optional)
;funcchar = * ; function lead-in character (defaults to '*')
;endchar = # ; command mode end character (defaults to '#')
;hangtime=1000 ; squelch tail hang time (in ms) (optional)
;totime=100000 ; transmit time-out time (in ms) (optional)
;idtime=30000 ; id interval time (in ms) (optional)
;politeid=30000 ; time in milliseconds before ID timer
; expires to try and ID in the tail.
; (optional, default is 30000).
;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none
;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none
;[002] ; Node ID of remote base
;rxchannel = Zap/5 ; Rx audio/signalling channel
; Note: if you use a unified interface (tx/rx on one channel), only
; specify the rxchannel and the txchannel will be assumed from the rxchannel
;txchannel = Zap/6 ; Tx audio/signalling channel
;functions = functions-remote
;remote = rbi ; Set remote=y for dumb remote or
; remote=rbi for Doug Hall RBI1
;iobase = 0x378 ; Specify IO port for parallel port (optional)
;[functions-repeater]
;1=ilink,1 ; Specific link disconnect
;2=ilink,2 ; Specific Link connect - monitor only
;3=ilink,3 ; Specific Link connect - transceive
;4=ilink,4 ; Enter command mode on a specific link
;7=ilink,5 ; Link status
;;XX=ilink,6 ; Disconnect all links (not used here)
;80=status,1 ; System info
;81=status,2 ; Time
;82=status,3 ; app_rpt.c Version
;6=autopatchup ; Autopatch up
;0=autopatchdn ; Autopatch down
;90=cop,1 ; System warm boot
;91=cop,2 ; System enable
;92=cop,3 ; System disable
;[functions-remote]
;0=remote,1 ; Retrieve Memory
;1=remote,2 ; Set freq.
;2=remote,3 ; Set Rx PL tone.
;40=remote,100 ; Rx PL off
;41=remote,101 ; Rx PL on
;42=remote,102 ; Tx PL off
;43=remote,103 ; Tx PL on
;44=remote,104 ; Low Pwr
;45=remote,105 ; Med Pwr
;46=remote,106 ; Hi Pwr
;5=remote,5 ; Status
;[telemetry]
; Telemetry entries are shared across all repeaters
; Can be a tone sequence, morse string, or a file
;
; |t - Tone escape sequence
;
; Tone sequences consist of 1 or more 4-tuple entries (freq1, freq2, duration, amplitude)
; Single frequencies are created by setting freq1 or freq2 to zero.
;
; |m - Morse escape sequence
;
; Sends Morse code at the telemetry amplitude and telemetry frequency as defined in the
; [morse] section.
;
; Follow with an alphanumeric string
;
; |i - Morse ID escape sequence
;
; Sends Morse code at the ID amplitude and ID frequency as defined in the
; [morse] section.
;
; Follow with an alphanumeric string
;ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048)
;ct2=|t(660,880,150,2048)
;ct3=|t(440,0,150,2048)
;ct4=|t(550,0,150,2048)
;ct5=|t(660,0,150,2048)
;ct6=|t(880,0,150,2048)
;ct7=|t(660,440,150,2048)
;ct8=|t(700,1100,150,2048)
;remotetx=|t(2000,0,75,2048)(0,0,75,0)(1600,0,75,2048);
;remotemon=|t(1600,0,75,2048)
;cmdmode=|t(900,903,200,2048)
;functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048)
;[morse]
;speed=20 ; Approximate speed in WPM
;frequency=800 ; Morse Telemetry Frequency
;amplitude=4096 ; Morse Telemetry Amplitude
;idfrequency=330 ; Morse ID Frequency
;idamplitude=2048 ; Morse ID Amplitude
;[nodes]
;000 = context_A@1.2.3.4/1234
;001 = context_B@5.6.7.8/4321
;of course, you can also specify these with domain names, but why rely
;on DNS working unnecessarily?
;[memory]
;; this example gives you 146.460, simplex, 100.0 HZ PL, hi-power, transmit PL
;00 = 146.460,100.0,sht
;; this example gives you 146.940, minus offset, 100.0 HZ PL, low-power, no PL
;01 = 146.940,100.0,-l
; The format for these entries is: Receive-Freq,Receive-PL,Attrbutes
; Attributes: l=low power, m=medium power, h=high power, -=minus offset,
; s=simplex, +=plus offset, t=tx PL enable, r=rx PL enable
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
; say.conf
;
; language configuration
;
; The new language routines produce strings of the form
; prefix:[format:]data
; that are matched against the rules in this file to produce
; an output.
;
; The data is generally the string to be spelled (either a number,
; an ascii string or a date/time in the format specified below).
; It is available, in the right hand side of a rule, as variable ${SAY}.
;
; The format is optional and normally used only for date/time.
; The prefix is used to select the pronunciation - standard
; prefixes are
; num used for numbers
; enum used for enumerations
; date for dates
; time for times
; datetime for dates and times
; char for character strings
; phonetic for phonetic strings
; digit for digit strings
;
; but others can be used at will.
;
; Processing occurs as follows:
; If the format is empty, or there is no format, the entire
; string is matched against one of the pattern on the left hand side.
; On the first match, the various comma-separated components on the right
; hand side are pronounced, as follows:
; + a component starting with a prefix: (i.e. with a ':' in it)
; is re-processed according to these rules;
; + a component without a ':' in it is considered a filename and
; the corresponding file is played.
;
; If the format is non-empty, the format is split into its components
; (individual characters, or filenames in single quotes), and then
; filenames are played, whereas single characters are used to
; generate a new string format:pat:data to be processed.
;
; DATES/AND TIMES assume that the date info is available in
; the form YYYYMMDDHHmm.ss-dow-doy
; with 4 digits for the year, 2 for month, day, hour, minutes, seconds,
; one digit for the day-of-week, and 3 digits for the day-of-year.
;
; Example:
; datetime::200604172030.00-4-102
; (typical format for a date) is first matched against the line
; datetime::. => date:AdBY 'digits/at' IMp:${SAY}
; which is normally present with the default format for dates.
; In turn, the format string "AdBY 'digits/at' IMp" results in
; the sequence
; date:A:200604172030.00-4-102
; date:d:200604172030.00-4-102
; date:B:200604172030.00-4-102
; date:Y:200604172030.00-4-102
; digits/at
; date:I:200604172030.00-4-102
; date:M:200604172030.00-4-102
; date:p:200604172030.00-4-102
;
;
; Remember, normally X Z N are special, and the search is
; case insensitive, so you must use [X] [N] [Z] .. if you
; want exact match.
; We start with the basic rules that might be more-or-less
; language-independent
[digit-base](!) ; base rule for digit strings
; XXX incomplete yet
_digit:[0-9] => digits/${SAY}
_digit:[-] => letters/dash
_digit:[*] => letters/star
_digit:[@] => letters/at
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
[date-base](!) ; base rules for dates and times
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. => num:${SAY:0:4} ; year, 19xx
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
_date:[de]:. => num:${SAY:6:2} ; day of month
_date:[hH]:. => num:${SAY:8:2} ; hour
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[M]:. => num:${SAY:10:2} ; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
_date:[pP]:. => digits/p-m ; am pm
_date:[S]:. => num:${SAY:13:2} ; seconds
[en-base](!)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
; enumeration
_e[n]um:X => digits/h-${SAY}
_e[n]um:1X => digits/h-${SAY}
_e[n]um:[2-9]0 => digits/h-${SAY}
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
[it](digit-base,date-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[en](en-base,date-base,digit-base)
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[de](date-base,digit-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
;
;
;
;
; begin sip 004a
;
;
;
;
; Contexto general
[general]
port = 5060 ; Puerto en el que empezamos
bindaddr = 0.0.0.0 ; dirección o direcciones ip 0.0.0.0 = todas
context=oficina ; Contexto default para todos
tos=lowdelay
dtmfmode=rfc2833 ; info ; Tonos DTMF
disallow=all ; Deshabilita todos los codecs
allow=ulaw ; Permite el codec ulaw (g711) 10kb/s
allow=ilbc ; Permite el codec ilbc 3kb/s
allow=gsm ; Permite el codec gsm 3kb/s
allow=g729 ; Permite el codec g729 2.5kb/s (propietario)
; Hacemos login en FWD (registrando) para recibir llamadas a nuestro numero y enviarlas
; A la extensión 21
; FWD number 77443 pointing to extension 21
; register => 77443:miclave@fwd.pulver.com/21
; Para poder sacar llamadas por FWD
; FWD account
;[fwd.pulver.com]
; type=peer
; host=fwd.pulver.com
; fromuser=77443
; fromdomain=fwd.pulver.com
; username=77443
; secret=miclave
; dtmfmode=rfc2833
; Extension 500
[sip500]
type=friend
secret=ext500
context=oficina
callerid="Nombre Apellido" <500>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=500@default,500
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
; Extension 501
[sip501]
type=friend
secret=ext501
context=oficina
callerid="Nombre Apellido" <501>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=501@default,501
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
; Nombre 502
; Extension 502
[sip502]
type=friend
secret=ext502
context=oficina
callerid="Nombre correo" <502>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=502@default,502
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
c
; Nombre 503
; Extension 503
[sip503]
type=friend
secret=ext503
context=oficina
callerid="Nombre correo" <503>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=503@default,503
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
; Nombre 553
; Extension 553
[sip553]
type=friend
secret=ext553
context=oficina
callerid="Nombre correo" <553>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=553@default,553
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
;[sip502]
;type=friend
;callerid="JetAccess Remote ATA" <502>
;host=dynamic ; This device needs to register
;username=sip502
;secret=ext502
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
[sip510]
type=friend
context=oficina
callerid="Nombre Apellido" <510>
username=sip510
secret=ext510
nat=no
canreinvite=yes
dtmfmode=info
mailbox=510@default
disallow=all
allow=ulaw
allow=alaw
allow=g729
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
[sip550]
type=friend
callerid="XLite Apellido remote" <550>
host=dynamic ; This device needs to register
username=sip550
secret=ext550
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
[polycom-check-cfg]
Event=>check-sync
Content-Length=>0
; Untested
[sipura-check-cfg]
Event=>resync
Content-Length=>0
; Untested
[grandstream-check-cfg]
Event=>sys-control
; Untested
[cisco-check-cfg]
Event=>check-sync
Content-Length=>0
; Untested - from Snom docs
[reboot-snom]
Event=>reboot
Content-Length=>0
;
; Skinny Configuration for Asterisk
;
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 0.0.0.0 ; Address to bind to
dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
; allow = all
; disallow =
; Typical config for 12SP+
;[florian]
;device=SEP00D0BA847E6B
;model=12SP ; Specific model of device, for button templates
; Valid models: 12SP, 30VIP, 7910, 7920 (so far)
;version=P002G204 ; Thanks critch
;context=did
;line => 120 ; Dial(Skinny/120@florian)
; Typical config for a 7910
;[duba] ; Device name
;model=7910 ; Device model
;device=SEP0007EB463101 ; Offical identifier
;version=P002F202 ; Firmware version identifier
;host=192.168.1.144 ;
;permit=192.168.0/24 ; Optional, used for authentication
;nat=0
;callerid="George W. Bush" <202-456-1414>
;mailbox=500
;callwaiting=1
;transfer=1
;threewaycalling=1
;context=default
;line => 500 ; Dial(Skinny/500@duba)
; Typical config for a 7940 / ATA
;[support]
;device=SEP0007EB463121
;nat=0
;callerid="Customer Support" <810-234-1212>
;mailbox=100
;context=inbound
;linelabel="Support Line" ; Displays next to the line button on 7940's and 7960s
;line => 100
;callerid="John Chambers" <408-526-4000>
;context=did
;linelabel="John"
;mailbox=110
;line => 110
;
; Configuration for Shared Line Appearances (SLA).
;
; See doc/sla.pdf for more information.
;
; ---- General Options ----------------
[general]
;attemptcallerid=no ; Attempt CallerID handling. The default value for this
; is "no" because CallerID handling with an SLA setup is
; known to not work properly in some situations. However,
; feel free to enable it if you would like. If you do, and
; you find problems, please do not report them.
; -------------------------------------
; ---- Trunk Declarations -------------
;
;[line1] ; Provide a name for this trunk.
;type=trunk ; This line is what marks this entry as a trunk.
;device=Zap/3 ; Map this trunk declaration to a specific device.
; NOTE: You can not just put any type of channel here.
; Zap channels can be directly used. IP trunks
; require some indirect configuration which is
; described in doc/sla.pdf.
;autocontext=line1 ; This supports automatic generation of the dialplan entries
; if the autocontext option is used. Each trunk should have
; a unique context name. Then, in zapata.conf, this device
; should be configured to have incoming calls go to this context.
;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging
; it up as an unanswered call. The value is in seconds.
;barge=no ; If this option is set to "no", then no station will be
; allowed to join a call that is in progress. The default
; value is "yes".
;hold=private ; This option configure hold permissions for this trunk.
; "open" - This means that any station can put this trunk
; on hold, and any station can retrieve it from
; hold. This is the default.
; "private" - This means that once a station puts the
; trunk on hold, no other station will be
; allowed to retrieve the call from hold.
;[line2]
;type=trunk
;device=Zap/4
;autocontext=line2
;[line3]
;type=trunk
;device=Zap/3
;autocontext=line3
;[line4]
;type=trunk
;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa
; application can be used to support IP trunks.
; See doc/sla.pdf on more information on how
; IP trunks work.
;autocontext=line4
; --------------------------------------
; ---- Station Declarations ------------
;[station1] ; Define a name for this station.
;type=station ; This line indicates that this entry is a station.
;device=SIP/station1 ; Each station must be mapped to a device.
;autocontext=sla_stations ; This supports automatic generation of the dialplan entries if
; the autocontext option is used. All stations can use the same
; context without conflict. The device for this station should
; have its context configured to the same one listed here.
;ringtimeout=10 ; Set a timeout for how long to allow the station to ring for an
; incoming call, in seconds.
;ringdelay=10 ; Set a time for how long to wait before beginning to ring this station
; once there is an incoming call, in seconds.
;hold=private ; This option configure hold permissions for this station. Note
; that if private hold is set in the trunk entry, that will override
; anything here. However, if a trunk has open hold access, but this
; station is set to private hold, then the private hold will be in
; effect.
; "open" - This means that once this station puts a call
; on hold, any other station is allowed to retrieve
; it. This is the default.
; "private" - This means that once this station puts a
; call on hold, no other station will be
; allowed to retrieve the call from hold.
;trunk=line1 ; Individually list all of the trunks that will appear on this station. This
; order is significant. It should be the same order as they appear on the
; phone. The order here defines the order of preference that the trunks will
; be used.
;trunk=line2
;trunk=line3,ringdelay=5 ; A ring delay for the station can also be specified for a specific trunk.
; If a ring delay is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
;trunk=line4,ringtimeout=5 ; A ring timeout for the station can also be specified for a specific trunk.
; If a ring timeout is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
;[station](!) ; When there are a lot of stations that are configured the same way,
; it is convenient to use a configuration template like this so that
; the common settings stay in one place.
;type=station
;autocontext=sla_stations
;trunk=line1
;trunk=line2
;trunk=line3
;trunk=line4
;[station2](station) ; Define a station that uses the configuration from the template "station".
;device=SIP/station2
;
;[station3](station)
;device=SIP/station3
;
;[station4](station)
;device=SIP/station4
;
;[station5](station)
;device=SIP/station5
; --------------------------------------
; Asterisk SMDI configuration
[interfaces]
; Specify serial ports to listen for SMDI messages on below. These will be
; referenced later in zapata.conf. If you do not specify any interfaces then
; SMDI will be disabled. Interfaces can have several different attributes
; associated with them.
; Set the number of stop bits to use per character here. The default is no,
; in which case one stop bit will be used.
;twostopbits = no
; Character size or bit length is the size of each character sent across the
; link. Character size can be 7 or 8. The default is 7.
;charsize = 7
; If you need parity checking enabled you can turn it on here. Acceptable
; values are even, odd, and none. The default is even.
;paritybit = even
; The baudrate to use for this port. Acceptable values are 1200, 2400, 4800,
; and 9600. The default is 9600.
;baudrate = 1200
; Often the numbering scheme for a set of mailboxes or extensions will not be 7
; or 10 digits (as SMDI requires). Use the msdstrip option to strip unused
; digits from the start of numbers.
;msdstrip = 0
; Occasionally Asterisk and the SMDI switch may become out of sync. If this
; happens, Asterisk will appear one or several calls behind as it processes
; voicemail requests. To prevent this from happening, adjust the msgexpirytime.
; This will make Asterisk discard old SMDI messages that have not yet been
; processed. The default expiry time is 30000 milliseconds.
;msgexpirytime = 30000
;smdiport => /dev/ttyS0
;
; UDPTL Configuration (UDPTL is one of the transports for T.38)
;
[general]
;
; UDPTL start and UDPTL end configure start and end addresses
;
udptlstart=4000
udptlend=4999
;
; Whether to enable or disable UDP checksums on UDPTL traffic
;
;udptlchecksums=no
;
; The error correction type to be sent
;
T38FaxUdpEC = t38UDPFEC
;T38FaxUdpEC = t38UDPRedundancy
;
; The maximum length of a UDPTL packet
;
T38FaxMaxDatagram = 400
;
; The number of error correction entries in a UDPTL packet
;
udptlfecentries = 3
;
; The span over which parity is calculated for FEC in a UDPTL packet
;
udptlfecspan = 3
;
; User configuration
;
; Creating entries in users.conf is a "shorthand" for creating individual
; entries in each configuration file. Using users.conf is not intended to
; provide you with as much flexibility as using the separate configuration
; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the
; simple task of adding users. Note that creating individual items (e.g.
; custom SIP peers, IAX friends, etc.) will allow you to override specific
; parameters within this file. Parameter names here are the same as they
; appear in the other configuration files. There is no way to change the
; value of a parameter here for just one subsystem.
;
[general]
;
; Full name of a user
;
fullname = New User
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Set voicemail mailbox 6000 password to 1234
;
vmsecret = 1234
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = yes
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
;[6000]
;fullname = Joe User
;email = joe@foo.bar
;secret = 1234
;zapchan = 1
;hasvoicemail = yes
;vmsecret = 1234
;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
;callwaiting = no
;context = international
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
format=wav ;format=g723sf|wav49|wav
serveremail=soporte@mailinator.net ; Who the e-mail notification should appear to come from
attach=no ; Should the email contain the voicemail as an attachment
;maxmessage=180 ; Maximum length of a voicemail message in seconds
;minmessage=3 ; Minimum length of a voicemail message in seconds
;maxgreet=60 ; Maximum length of greetings in seconds
skipms=3000 ; How many miliseconds to skip forward/back when rew/ff in message playback
maxsilence=2 ; How many seconds of silence before we end the recording
silencethreshold=30 ; Silence threshold (what we consider silence, the lower, the more sensitive)
maxlogins=3 ; Max number of failed login attempts
languaje=es
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp
;directoryintro=dir-intro ; For the directory, you can override the intro file if you want
;charset=ISO-8859-1 ; The character set for voicemail messages can be specified here
;adsifdn=0000000F ; The ADSI feature descriptor number to download to
;adsisec=9BDBF7AC ; The ADSI security lock code
;adsiver=1 ; The ADSI voicemail application version number.
;pbxskip=yes ; Skip the "[PBX]:" string from the message title
;fromstring=The Asterisk PBX ; Change the From: string
;usedirectory=yes ; Permit finding entries for forward/compose from the directory
;pagerfromstring=The Asterisk PBX ;Change the From: string for pager messages
; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can be up to 512 characters due to a limitation in
; asterisk config files.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default shows just
; the CIDNAME, if it is not null, else just the CIDNUM, or "an unknown caller" if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
; You can override the default program to send e-mail if you wish, too
;mailcmd=/usr/sbin/sendmail -t
; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the general section, and overriden in the per-mailbox
; settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no.
attach=yes ; Attach the voicemail to the notification email *NOT* the pager email
;saycid=yes ; Say the caller id information before the message. If not described,
; or set to no, it will be in the envelope
; cidinternalcontexts=intern ; Internal Context for Name Playback instead of extension digits when saying caller id.
; sayduration=no ; Turn on/off the duration information before the message. [ON by default]
; saydurationm=2 ; Specify the minimum duration to say. Default is 2 minutes
; dialout=fromvm ; Context to dial out from [option 4 from the advanced menu]
; if not listed, dialing out will not be permitted
sendvoicemail=yes ; Context to Send voicemail from [option 5 from the advanced menu]
; if not listed, sending messages from inside voicemail will not be
; permitted
; callback=fromvm ; Context to call back from
; if not listed, calling the sender back will not be permitted
; review=yes ; Allow sender to review/rerecord their message before saving it [OFF by default
; operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to
; reach an operator [OFF by default]
; envelope=no ; Turn on/off envelope playback before message playback. [ON by default]
; This does NOT affect option 3,3 from the advanced options menu
; delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only]
; This is intended for use with users who wish to receive their voicemail ONLY by email.
; nextaftercmd=yes ; Skips to the next message after hitting 7 or 9 to delete/save current message.
; [global option only at this time]
; forcename=yes ; Forces a new user to record their name. A new user is
; determined by the password being the same as
; the mailbox number. The default is "no".
; forcegreetings=no ; This is the same as forcename, except for recording
; greetings. The default is "no".
; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory
; The default is "no".
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M 'hours'
[default]
500 => 500,Nombre Apellido,correo@mailinator.com,,attach=yes
501 => 501,Nombre Apellido,correo@mailinator.com,,attach=yes
502 => 502,Nombre correo,correo@mailinator.net,,attach=yes
503 => 503,Nombre correo,correo@mailinator.net,,attach=yes
553 => 553,Nombre correo,correo@mailinator.net,,attach=yes
510 => 510,Nombre Apellido,correo@mailinator.com,,attach=yes
550 => 550,Nombre Apellido remote,correo@mailinator.com,attach=yes
; V6PCI/V12PCI config file for VoiceTronix Hardware
; Options
; For [general] section
; type = v12pci|v6pci|v4pci
; cards = number of cards
; indication = 1 ( To use Asterisk indication tones)
; ecsuppthres = 0|2048|4096 (none,-24db,-18db only for use with OpenLine4)
; dtmfidd = 3000 (Inter Digit Delay timeout for when collecting DTMF tones for dialling from a Station port, in ms)
; ast-dtmf-det=1 ( To use Asterisk DTMF detection )
; relaxdtmf=1 ( Used with ast-dtmf-det )
; break-for-dtmf=no (When a native bridge occurs between 2 vpb channels, it will only break the connection for '#' and '*')
; timer_period_ring=4000 (Set the maximum period between received rings, default 4000ms)
;
; For [interface] section
; board = board_number (1, 2, 3, ...)
; channel = channel_number (1,2,3...)
; mode = fxo|immediate|dialtone -- for type of line and line handling
; context = starting context
; echocancel = on|off (on by default of v4pci, off by default for others)
; callerid = on|off|v23|bell (on => to collect caller ID if available between 1st/2nd rings using vpb functions)
; (v23|bell => collect caller ID using asterisk functions)
; Or for use with FXS channels a '"name" <location>' format can be used to set the channels CID
;
; UseLoopDrop = 0|1 (enables the use of Loop Drop detection, on by default in
; some cases spurious loop-drops can cause unexpected
; hangup detection)
;
; Gain settings
; txgain => Transmit Software Gain (-12 => 12)
; rxgain => Receive Software Gain (-12 => 12)
; txhwgain => Transmit hardware gain (-12 => 12)
; rxhwgain => Receive Hardware gain (-12 => 12)
;
; These are advanced settings and only mentioned for fullnes.
; bal1 => Hybrid balance codec register 1
; bal2 => Hybrid balance codec register 2
; bal3 => Hybrid balance codec register 3
;
; Dial translations - if you want a pause or hook-flash in your dial string
; you can use "w" for pause (wait) or "f" for "hook-flash", eg:
; exten => _9XXX,1,Dial(vpb/g1/ww${EXTEN:${TRUNKMSD}})
;
;
[general]
type = v12pci
;type = v6pci
;type = v4pci
cards = 1
[interfaces]
board = 1
echocancel = on
; For OpenLine4 cards
;context = demo
;mode = fxo
;channel = 1
;channel = 2
;channel = 3
;channel = 4
; For OpenSwith12 with jumpers at factory default
context = demo
mode = fxo
channel = 9
channel = 10
channel = 11
channel = 12
context = local
mode = dialtone
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7
channel = 8
;
; For OpenSwitch6
; Note that V6PCI channel numbers start at 7!
;context = demo
;mode = fxo
;channel = 7
;channel = 8
;mode = dialtone
;channel = 9
;channel = 10
;channel = 11
;channel = 12
[channels]
busydetect=no
busycount=5
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
callerid=asreceived
callprogress=no
musiconhold=yes
channel => 1-2