diff options
Diffstat (limited to '1.2-netsec/configs')
47 files changed, 0 insertions, 5358 deletions
diff --git a/1.2-netsec/configs/adsi.conf.sample b/1.2-netsec/configs/adsi.conf.sample deleted file mode 100644 index 0f36f80da..000000000 --- a/1.2-netsec/configs/adsi.conf.sample +++ /dev/null @@ -1,8 +0,0 @@ -; -; Sample ADSI Configuration file -; -[intro] -alignment = center -greeting => Welcome to the -greeting => Asterisk -greeting => Open Source PBX diff --git a/1.2-netsec/configs/adtranvofr.conf.sample b/1.2-netsec/configs/adtranvofr.conf.sample deleted file mode 100644 index dc7bcfc7c..000000000 --- a/1.2-netsec/configs/adtranvofr.conf.sample +++ /dev/null @@ -1,39 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/agents.conf.sample b/1.2-netsec/configs/agents.conf.sample deleted file mode 100644 index c47100b18..000000000 --- a/1.2-netsec/configs/agents.conf.sample +++ /dev/null @@ -1,80 +0,0 @@ -[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 diff --git a/1.2-netsec/configs/alarmreceiver.conf.sample b/1.2-netsec/configs/alarmreceiver.conf.sample deleted file mode 100644 index bf767dea3..000000000 --- a/1.2-netsec/configs/alarmreceiver.conf.sample +++ /dev/null @@ -1,80 +0,0 @@ -; -; 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 -; diff --git a/1.2-netsec/configs/alsa.conf.sample b/1.2-netsec/configs/alsa.conf.sample deleted file mode 100644 index 98819250b..000000000 --- a/1.2-netsec/configs/alsa.conf.sample +++ /dev/null @@ -1,31 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/asterisk.adsi b/1.2-netsec/configs/asterisk.adsi deleted file mode 100644 index a275502ac..000000000 --- a/1.2-netsec/configs/asterisk.adsi +++ /dev/null @@ -1,159 +0,0 @@ -; -; Asterisk default ADSI script -; -; -; Begin with the preamble requirements -; -DESCRIPTION "Asterisk PBX" ; Name of vendor -VERSION 0x00 ; Version of stuff -;SECURITY "_AST" ; Security code -SECURITY 0X9BDBF7AC ; Security code -FDN 0x0000000F ; Descriptor number - -; -; Flags -; -FLAG "nocallwaiting" - -; -; Predefined strings -; -DISPLAY "titles" IS "** Asterisk PBX **" -DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT -DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT -DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT -DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT -DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT -DISPLAY "callended" IS "Call ended." JUSTIFY LEFT -DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT -DISPLAY "busy" IS "Busy." JUSTIFY LEFT -DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT -DISPLAY "cwdisabled" IS "Callwait disabled" -DISPLAY "empty" IS "asdf" - -; -; Begin soft key definitions -; -KEY "callfwd" IS "CallFwd" OR "Call Forward" - OFFHOOK - VOICEMODE - WAITDIALTONE - SENDDTMF "*60" - GOTO "offHook" -ENDKEY - -KEY "vmail_OH" IS "VMail" OR "Voicemail" - OFFHOOK - VOICEMODE - WAITDIALTONE - SENDDTMF "8500" -ENDKEY - -KEY "vmail" IS "VMail" OR "Voicemail" - SENDDTMF "8500" -ENDKEY - -KEY "backspace" IS "BackSpc" OR "Backspace" - BACKSPACE -ENDKEY - -KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait" - SENDDTMF "*70" - SETFLAG "nocallwaiting" - SHOWDISPLAY "cwdisabled" AT 4 - TIMERCLEAR - TIMERSTART 1 -ENDKEY - -KEY "cidblock" IS "CIDBlk" OR "Block Callerid" - SENDDTMF "*67" - SETFLAG "nocallwaiting" -ENDKEY - -; -; Begin main subroutine -; - -SUB "main" IS - IFEVENT NEARANSWER THEN - CLEAR - SHOWDISPLAY "titles" AT 1 NOUPDATE - SHOWDISPLAY "talkingto" AT 2 NOUPDATE - SHOWDISPLAY "callname" AT 3 - SHOWDISPLAY "callnum" AT 4 - GOTO "stableCall" - ENDIF - IFEVENT OFFHOOK THEN - CLEAR - CLEARFLAG "nocallwaiting" - CLEARDISPLAY - SHOWDISPLAY "titles" AT 1 - SHOWKEYS "vmail" - SHOWKEYS "cidblock" - SHOWKEYS "cwdisable" UNLESS "nocallwaiting" - GOTO "offHook" - ENDIF - IFEVENT IDLE THEN - CLEAR - SHOWDISPLAY "titles" AT 1 - SHOWKEYS "vmail_OH" - ENDIF - IFEVENT CALLERID THEN - CLEAR -; SHOWDISPLAY "titles" AT 1 NOUPDATE -; SHOWDISPLAY "incoming" AT 2 NOUPDATE - SHOWDISPLAY "callname" AT 3 NOUPDATE - SHOWDISPLAY "callnum" AT 4 - ENDIF - IFEVENT RING THEN - CLEAR - SHOWDISPLAY "titles" AT 1 NOUPDATE - SHOWDISPLAY "incoming" AT 2 - ENDIF - IFEVENT ENDOFRING THEN - SHOWDISPLAY "missedcall" AT 2 - CLEAR - SHOWDISPLAY "titles" AT 1 - SHOWKEYS "vmail_OH" - ENDIF - IFEVENT TIMER THEN - CLEAR - SHOWDISPLAY "empty" AT 4 - ENDIF -ENDSUB - -SUB "offHook" IS - IFEVENT FARRING THEN - CLEAR - SHOWDISPLAY "titles" AT 1 NOUPDATE - SHOWDISPLAY "ringing" AT 2 NOUPDATE - SHOWDISPLAY "callname" at 3 NOUPDATE - SHOWDISPLAY "callnum" at 4 - ENDIF - IFEVENT FARANSWER THEN - CLEAR - SHOWDISPLAY "talkingto" AT 2 - GOTO "stableCall" - ENDIF - IFEVENT BUSY THEN - CLEAR - SHOWDISPLAY "titles" AT 1 NOUPDATE - SHOWDISPLAY "busy" AT 2 NOUPDATE - SHOWDISPLAY "callname" at 3 NOUPDATE - SHOWDISPLAY "callnum" at 4 - ENDIF - IFEVENT REORDER THEN - CLEAR - SHOWDISPLAY "titles" AT 1 NOUPDATE - SHOWDISPLAY "reorder" AT 2 NOUPDATE - SHOWDISPLAY "callname" at 3 NOUPDATE - SHOWDISPLAY "callnum" at 4 - ENDIF -ENDSUB - -SUB "stableCall" IS - IFEVENT REORDER THEN - SHOWDISPLAY "callended" AT 2 - ENDIF -ENDSUB - diff --git a/1.2-netsec/configs/cdr.conf.sample b/1.2-netsec/configs/cdr.conf.sample deleted file mode 100644 index 331b6ed9a..000000000 --- a/1.2-netsec/configs/cdr.conf.sample +++ /dev/null @@ -1,51 +0,0 @@ -; -; 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 - diff --git a/1.2-netsec/configs/cdr_custom.conf.sample b/1.2-netsec/configs/cdr_custom.conf.sample deleted file mode 100644 index 4af17c37b..000000000 --- a/1.2-netsec/configs/cdr_custom.conf.sample +++ /dev/null @@ -1,6 +0,0 @@ -; -; 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)}" - diff --git a/1.2-netsec/configs/cdr_manager.conf.sample b/1.2-netsec/configs/cdr_manager.conf.sample deleted file mode 100644 index 1d7984ba4..000000000 --- a/1.2-netsec/configs/cdr_manager.conf.sample +++ /dev/null @@ -1,6 +0,0 @@ -; -; Asterisk Call Management CDR -; -[general] -enabled = no - diff --git a/1.2-netsec/configs/cdr_odbc.conf.sample b/1.2-netsec/configs/cdr_odbc.conf.sample deleted file mode 100644 index 6245e37eb..000000000 --- a/1.2-netsec/configs/cdr_odbc.conf.sample +++ /dev/null @@ -1,12 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/cdr_pgsql.conf.sample b/1.2-netsec/configs/cdr_pgsql.conf.sample deleted file mode 100644 index 0784c7b08..000000000 --- a/1.2-netsec/configs/cdr_pgsql.conf.sample +++ /dev/null @@ -1,9 +0,0 @@ -; Sample Asterisk config file for CDR logging to PostgresSQL - -[global] -;hostname=localhost -;port=5432 -;dbname=asterisk -;password=password -;user=postgres -;table=cdr ;SQL table where CDRs will be inserted diff --git a/1.2-netsec/configs/cdr_tds.conf.sample b/1.2-netsec/configs/cdr_tds.conf.sample deleted file mode 100644 index 9fffec099..000000000 --- a/1.2-netsec/configs/cdr_tds.conf.sample +++ /dev/null @@ -1,9 +0,0 @@ -; Sample Asterisk config file for CDR logging to FreeTDS - -;[global] -;hostname=fs.malico.loc -;port=1433 -;dbname=MalicoHN -;user=mangUsr -;password= -;charset=BIG5 diff --git a/1.2-netsec/configs/codecs.conf.sample b/1.2-netsec/configs/codecs.conf.sample deleted file mode 100644 index c8caeab60..000000000 --- a/1.2-netsec/configs/codecs.conf.sample +++ /dev/null @@ -1,65 +0,0 @@ -[speex] -; CBR encoding quality [0..10] -; used only when vbr = false -quality => 3 - -; codec complexity [0..10] -; tradeoff between cpu/quality -complexity => 2 - -; perceptual enhancement [true / false] -; improves clarity of decoded speech -enhancement => true - -; voice activity detection [true / false] -; reduces bitrate when no voice detected, used only for CBR -; (implicit in VBR/ABR) -vad => true - -; variable bit rate [true / false] -; uses bit rate proportionate to voice complexity -vbr => true - -; available bit rate [bps, 0 = off] -; encoding quality modulated to match this target bit rate -; not recommended with dtx or pp_vad - may cause bandwidth spikes -abr => 0 - -; VBR encoding quality [0-10] -; floating-point values allowed -vbr_quality => 4 - -; discontinuous transmission [true / false] -; stops transmitting completely when silence is detected -; pp_vad is far more effective but more CPU intensive -dtx => false - -; preprocessor configuration -; these options only affect Speex v1.1.8 or newer - -; enable preprocessor [true / false] -; allows dsp functionality below but incurs CPU overhead -preprocess => false - -; preproc voice activity detection [true / false] -; more advanced equivalent of DTX, based on voice frequencies -pp_vad => false - -; preproc automatic gain control [true / false] -pp_agc => false -pp_agc_level => 8000 - -; preproc denoiser [true / false] -pp_denoise => false - -; preproc dereverb [true / false] -pp_dereverb => false -pp_dereverb_decay => 0.4 -pp_dereverb_level => 0.3 - - -[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 diff --git a/1.2-netsec/configs/dnsmgr.conf.sample b/1.2-netsec/configs/dnsmgr.conf.sample deleted file mode 100644 index e34dbcf0a..000000000 --- a/1.2-netsec/configs/dnsmgr.conf.sample +++ /dev/null @@ -1,5 +0,0 @@ -[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)
\ No newline at end of file diff --git a/1.2-netsec/configs/dundi.conf.sample b/1.2-netsec/configs/dundi.conf.sample deleted file mode 100644 index a3c8c77d9..000000000 --- a/1.2-netsec/configs/dundi.conf.sample +++ /dev/null @@ -1,239 +0,0 @@ -; -; 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 - -;[*] -; diff --git a/1.2-netsec/configs/enum.conf.sample b/1.2-netsec/configs/enum.conf.sample deleted file mode 100644 index 8d7054a24..000000000 --- a/1.2-netsec/configs/enum.conf.sample +++ /dev/null @@ -1,22 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/extconfig.conf.sample b/1.2-netsec/configs/extconfig.conf.sample deleted file mode 100644 index 1cf923fb3..000000000 --- a/1.2-netsec/configs/extconfig.conf.sample +++ /dev/null @@ -1,51 +0,0 @@ -; -; 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 - diff --git a/1.2-netsec/configs/extensions.ael.sample b/1.2-netsec/configs/extensions.ael.sample deleted file mode 100644 index 87fe58039..000000000 --- a/1.2-netsec/configs/extensions.ael.sample +++ /dev/null @@ -1,62 +0,0 @@ -// -// Example AEL config file -// - -macro std-exten-ael( ext , dev ) { - Dial(${dev}/${ext},20); - switch(${DIALSTATUS}) { - case BUSY: - Voicemail(b${ext}); - break; - default: - Voicemail(u${ext}); - }; - catch a { - VoiceMailMain(${ext}); - return; - }; -}; - -context ael-demo { - s => { - Wait(1); - Answer(); - TIMEOUT(digit)=5; - TIMEOUT(response)=10; -restart: - Background(demo-congrats); -instructions: - for (x=0; ${x} < 3; x=${x} + 1) { - Background(demo-instruct); - WaitExten(); - }; - }; - 2 => { - Background(demo-moreinfo); - goto s|instructions; - }; - 3 => { - LANGUAGE()=fr; - goto s|restart; - }; - 500 => { - Playback(demo-abouttotry); - Dial(IAX2/guest@misery.digium.com); - Playback(demo-nogo); - goto s|instructions; - }; - 600 => { - Playback(demo-echotest); - Echo(); - Playback(demo-echodone); - goto s|instructions; - }; - _1234 => &std-exten-ael(${EXTEN}, "IAX2"); - # => { - Playback(demo-thanks); - Hangup(); - }; - t => jump #; - i => Playback(invalid); -}; - diff --git a/1.2-netsec/configs/extensions.conf.sample b/1.2-netsec/configs/extensions.conf.sample deleted file mode 100644 index d773cbbc3..000000000 --- a/1.2-netsec/configs/extensions.conf.sample +++ /dev/null @@ -1,492 +0,0 @@ -; -; Static extension configuration file, used by -; the pbx_config module. This is where you configure all your -; inbound and outbound calls in Asterisk. -; -; This configuration file is reloaded -; - With the "extensions reload" command in the CLI -; - With the "reload" command (that reloads everything) in the CLI - -; -; The "General" category is for certain variables. -; -[general] -; -; If static is set to no, or omitted, then the pbx_config will rewrite -; this file when extensions are modified. Remember that all comments -; made in the file will be lost when that happens. -; -; XXX Not yet implemented XXX -; -static=yes -; -; if static=yes and writeprotect=no, you can save dialplan by -; CLI command 'save dialplan' too -; -writeprotect=no -; -; If autofallthrough is set, then if an extension runs out of -; things to do, it will terminate the call with BUSY, CONGESTION -; or HANGUP depending on Asterisk's best guess (strongly recommended). -; -; If autofallthrough is not set, then if an extension runs out of -; things to do, asterisk will wait for a new extension to be dialed -; (this is the original behavior of Asterisk 1.0 and earlier). -; -autofallthrough=yes -; -; If clearglobalvars is set, global variables will be cleared -; and reparsed on an extensions reload, or Asterisk reload. -; -; If clearglobalvars is not set, then global variables will persist -; through reloads, and even if deleted from the extensions.conf or -; one if its included files, will remain set to the previous value. -; -clearglobalvars=no -; -; If priorityjumping is set to 'yes', then applications that support -; 'jumping' to a different priority based on the result of their operations -; will do so (this is backwards compatible behavior with pre-1.2 releases -; of Asterisk). Individual applications can also be requested to do this -; by passing a 'j' option in their arguments. -; -priorityjumping=no -; -; You can include other config files, use the #include command -; (without the ';'). Note that this is different from the "include" command -; that includes contexts within other contexts. The #include command works -; in all asterisk configuration files. -;#include "filename.conf" - -; The "Globals" category contains global variables that can be referenced -; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental -; variables, -; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid -; -[globals] -CONSOLE=Console/dsp ; Console interface for demo -;CONSOLE=Zap/1 -;CONSOLE=Phone/phone0 -IAXINFO=guest ; IAXtel username/password -;IAXINFO=myuser:mypass -TRUNK=Zap/g2 ; Trunk interface -; -; Note the 'g2' in the TRUNK variable above. It specifies which group (defined -; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in -; the specified group. The four possible options are: -; -; g: select the lowest-numbered non-busy Zap channel -; (aka. ascending sequential hunt group). -; G: select the highest-numbered non-busy Zap channel -; (aka. descending sequential hunt group). -; r: use a round-robin search, starting at the next highest channel than last -; time (aka. ascending rotary hunt group). -; R: use a round-robin search, starting at the next lowest channel than last -; time (aka. descending rotary hunt group). -; -TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) -;TRUNK=IAX2/user:pass@provider - -; -; Any category other than "General" and "Globals" represent -; extension contexts, which are collections of extensions. -; -; Extension names may be numbers, letters, or combinations -; thereof. If an extension name is prefixed by a '_' -; character, it is interpreted as a pattern rather than a -; literal. In patterns, some characters have special meanings: -; -; 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 -; -; For example the extension _NXXXXXX would match normal 7 digit dialings, -; while _1NXXNXXXXXX would represent an area code plus phone number -; preceeded by a one. -; -; Each step of an extension is ordered by priority, which must -; always start with 1 to be considered a valid extension. The priority -; "next" or "n" means the previous priority plus one, regardless of whether -; the previous priority was associated with the current extension or not. -; The priority "same" or "s" means the same as the previously specified -; priority, again regardless of whether the previous entry was for the -; same extension. Priorities may be immediately followed by a plus sign -; and another integer to add that amount (most useful with 's' or 'n'). -; Priorities may then also have an alias, or label, in -; parenthesis after their name which can be used in goto situations -; -; Contexts contain several lines, one for each step of each -; extension, which can take one of two forms as listed below, -; with the first form being preferred. One may include another -; context in the current one as well, optionally with a -; date and time. Included contexts are included in the order -; they are listed. -; -;[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. -; - -; -; Sample entries for extensions.conf -; -; -[dundi-e164-canonical] -; -; List canonical entries here -; -;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo) -;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7}) - -[dundi-e164-customers] -; -; If you are an ITSP or Reseller, list your customers here. -; -;exten => _12564286000,1,Dial(SIP/customer1) -;exten => _12564286001,1,Dial(IAX2/customer2) - -[dundi-e164-via-pstn] -; -; If you are freely delivering calls to the PSTN, list them here -; -;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428 -;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325 - -[dundi-e164-local] -; -; Context to put your dundi IAX2 or SIP user in for -; full access -; -include => dundi-e164-canonical -include => dundi-e164-customers -include => dundi-e164-via-pstn - -[dundi-e164-switch] -; -; Just a wrapper for the switch -; -switch => DUNDi/e164 - -[dundi-e164-lookup] -; -; Locally to lookup, try looking for a local E.164 solution -; then try DUNDi if we don't have one. -; -include => dundi-e164-local -include => dundi-e164-switch -; -; DUNDi can also be implemented as a Macro instead of using -; the Local channel driver. -; -[macro-dundi-e164] -; -; ARG1 is the extension to Dial -; -exten => s,1,Goto(${ARG1},1) -include => dundi-e164-lookup - -; -; Here are the entries you need to participate in the IAXTEL -; call routing system. Most IAXTEL numbers begin with 1-700, but -; there are exceptions. For more information, and to sign -; up, please go to www.gnophone.com or www.iaxtel.com -; -[iaxtel700] -exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) - -; -; The SWITCH statement permits a server to share the dialplain with -; another server. Use with care: Reciprocal switch statements are not -; allowed (e.g. both A -> B and B -> A), and the switched server needs -; to be on-line or else dialing can be severly delayed. -; -[iaxprovider] -;switch => IAX2/user:[key]@myserver/mycontext - -[trunkint] -; -; International long distance through trunk -; -exten => _9011.,1,Macro(dundi-e164,${EXTEN:4}) -exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) - -[trunkld] -; -; Long distance context accessed through trunk -; -exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1}) -exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) - -[trunklocal] -; -; Local seven-digit dialing accessed through trunk interface -; -exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) - -[trunktollfree] -; -; Long distance context accessed through trunk interface -; -exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) -exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) -exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) -exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) - -[international] -; -; Master context for international long distance -; -ignorepat => 9 -include => longdistance -include => trunkint - -[longdistance] -; -; Master context for long distance -; -ignorepat => 9 -include => local -include => trunkld - -[local] -; -; Master context for local, toll-free, and iaxtel calls only -; -ignorepat => 9 -include => default -include => parkedcalls -include => trunklocal -include => iaxtel700 -include => trunktollfree -include => iaxprovider -; -; You can use an alternative switch type as well, to resolve -; extensions that are not known here, for example with remote -; IAX switching you transparently get access to the remote -; Asterisk PBX -; -; switch => IAX2/user:password@bigserver/local -; -; An "lswitch" is like a switch but is literal, in that -; variable substitution is not performed at load time -; but is passed to the switch directly (presumably to -; be substituted in the switch routine itself) -; -; lswitch => Loopback/12${EXTEN}@othercontext -; -; An "eswitch" is like a switch but the evaluation of -; variable substitution is performed at runtime before -; being passed to the switch routine. -; -; eswitch => IAX2/context@${CURSERVER} - -[macro-stdexten]; -; -; Standard extension macro: -; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well -; ${ARG2} - Device(s) to ring -; -exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum -exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) - -exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; 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(b${ARG1}) ; If busy, send to voicemail w/ busy announce -exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start - -exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer - -exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain - -[macro-stdPrivacyexten]; -; -; Standard extension macro: -; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well -; ${ARG2} - Device(s) to ring -; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority) -; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)` -; -exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening) -exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) - -exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; 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(b${ARG1}) ; If busy, send to voicemail w/ busy announce -exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start - -exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite "Don't call again" script. - -exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script. - -exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer - -exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain - -[demo] -; -; We start with what to do when a call first comes in. -; -exten => s,1,Wait,1 ; Wait a second, just for fun -exten => s,n,Answer ; Answer the line -exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds -exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds -exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message -exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions -exten => s,n,WaitExten ; Wait for an extension to be dialed. - -exten => 2,1,BackGround(demo-moreinfo) ; Give some more information. -exten => 2,n,Goto(s,instruct) - -exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french -exten => 3,n,Goto(s,restart) ; Start with the congratulations - -exten => 1000,1,Goto(default,s,1) -; -; We also create an example user, 1234, who is on the console and has -; voicemail, etc. -; -exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." - ; (but skip if channel is not up) -exten => 1234,n,Macro(stdexten,1234,${CONSOLE}) - -exten => 1235,1,Voicemail(u1234) ; Right to voicemail - -exten => 1236,1,Dial(Console/dsp) ; Ring forever -exten => 1236,n,Voicemail(u1234) ; Unless busy - -; -; # for when they're done with the demo -; -exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" -exten => #,n,Hangup ; Hang them up. - -; -; A timeout and "invalid extension rule" -; -exten => t,1,Goto(#,1) ; If they take too long, give up -exten => i,1,Playback(invalid) ; "That's not valid, try again" - -; -; Create an extension, 500, for dialing the -; Asterisk demo. -; -exten => 500,1,Playback(demo-abouttotry); Let them know what's going on -exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo -exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site -exten => 500,n,Goto(s,6) ; Return to the start over message. - -; -; Create an extension, 600, for evaulating echo latency. -; -exten => 600,1,Playback(demo-echotest) ; Let them know what's going on -exten => 600,n,Echo ; Do the echo test -exten => 600,n,Playback(demo-echodone) ; Let them know it's over -exten => 600,n,Goto(s,6) ; Start over - -; -; Give voicemail at extension 8500 -; -exten => 8500,1,VoicemailMain -exten => 8500,n,Goto(s,6) -; -; Here's what a phone entry would look like (IXJ for example) -; -;exten => 1265,1,Dial(Phone/phone0,15) -;exten => 1265,n,Goto(s,5) - -;[mainmenu] -; -; Example "main menu" context with submenu -; -;exten => s,1,Answer -;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..." -;exten => s,n,WaitExten -;exten => 1,1,Goto(submenu,s,1) -;exten => 2,1,Hangup -;include => default -; -;[submenu] -;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback -;exten => s,n,Wait,2 -;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..." -;exten => s,n,WaitExten -;exten => 1,1,Goto(default,steve,1) -;exten => 2,1,Goto(default,mark,2) - -[default] -; -; By default we include the demo. In a production system, you -; probably don't want to have the demo there. -; -include => demo - -; -; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. -; Note that you must have a [sipprovider] section in sip.conf whereas -; the otherprovider.net example does not require such a peer definition -; -;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) -;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) - -; Real extensions would go here. Generally you want real extensions to be -; 4 or 5 digits long (although there is no such requirement) and start with a -; single digit that is fairly large (like 6 or 7) so that you have plenty of -; room to overlap extensions and menu options without conflict. You can alias -; them with names, too, and use global variables - -;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence -;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer -;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed -;exten => 6245,n,Voicemail(u6245) ; Voicemail (unavailable) -;exten => 6245,s+1,Hangup ; s+1, same as n -;exten => 6245,dial+101,Voicemail(b6245) ; Voicemail (busy) -;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit -;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14) -;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK} - -;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2 -;exten => mark,1,Goto(6275|1) ; alias mark to 6275 -;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil -;exten => wil,1,Goto(6236|1) -; -; Some other handy things are an extension for checking voicemail via -; voicemailmain -; -;exten => 8500,1,VoicemailMain -;exten => 8500,n,Hangup -; -; Or a conference room (you'll need to edit meetme.conf to enable this room) -; -;exten => 8600,1,Meetme(1234) -; -; Or playing an announcement to the called party, as soon it answers -; -;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg)) -; -; For more information on applications, just type "show applications" at your -; friendly Asterisk CLI prompt. -; -; 'show application <command>' will show details of how you -; use that particular application in this file, the dial plan. -; diff --git a/1.2-netsec/configs/features.conf.sample b/1.2-netsec/configs/features.conf.sample deleted file mode 100644 index 346d65192..000000000 --- a/1.2-netsec/configs/features.conf.sample +++ /dev/null @@ -1,32 +0,0 @@ -; -; 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 - -[applicationmap] -;testfeature => #9,callee,Playback,tt-monkeys ;Play tt-monkeys to - ;callee if #9 was pressed - diff --git a/1.2-netsec/configs/festival.conf.sample b/1.2-netsec/configs/festival.conf.sample deleted file mode 100644 index 774f1a16c..000000000 --- a/1.2-netsec/configs/festival.conf.sample +++ /dev/null @@ -1,35 +0,0 @@ -; -; 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 -; -; - diff --git a/1.2-netsec/configs/iax.conf.sample b/1.2-netsec/configs/iax.conf.sample deleted file mode 100644 index 26d637d8d..000000000 --- a/1.2-netsec/configs/iax.conf.sample +++ /dev/null @@ -1,418 +0,0 @@ - -; 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 Asterisk -; 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. -; Resyncing 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 -; -; Minimum and maximum amounts of time that IAX peers can request as -; a registration expiration interval (in seconds). -; minregexpire = 60 -; maxregexpire = 60 -; -; 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 - -;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list - ; just like friends added from the config file only on a - ; as-needed basis? (yes|no) - -;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) - ; If set to yes, when a IAX2 peer registers successfully, the ip address, - ; the origination port, the registration period, and the username of - ; the peer will be set to database via realtime. If not present, defaults to 'yes'. - -;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule - ; as if it had just registered? (yes|no|<seconds>) - ; If set to yes, 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 - ; registration interval. - -;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. (yes|no) - -; 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. -; -; -; NOTE: All hostnames and IP addresses in this file are for example purposes -; only; you should not expect any of them to actually be available for -; your use. -; -; -;[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 -;qualifysmoothing = yes ; use an average of the last two PONG - ; results to reduce falsly detected LAGGED hosts - ; Default: Off -;qualifyfreqok = 60000 ; how frequently to ping the peer when - ; everything seems to be ok, in milliseconds -;qualifyfreqnotok = 10000 ; how frequently to ping the peer when it's - ; either LAGGED or UNAVAILABLE, in milliseconds -;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 - diff --git a/1.2-netsec/configs/iaxprov.conf.sample b/1.2-netsec/configs/iaxprov.conf.sample deleted file mode 100644 index ad13166ed..000000000 --- a/1.2-netsec/configs/iaxprov.conf.sample +++ /dev/null @@ -1,83 +0,0 @@ -; -; 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 - diff --git a/1.2-netsec/configs/indications.conf.sample b/1.2-netsec/configs/indications.conf.sample deleted file mode 100644 index d70ac60ed..000000000 --- a/1.2-netsec/configs/indications.conf.sample +++ /dev/null @@ -1,611 +0,0 @@ -; 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. -; ringcadence = 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 be used automatically 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -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,!0/3000,!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 -ringcadence = 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 - -[es] -description = Spain -ringcadence = 1500,3000 -dial = 425 -busy = 425/200,0/200 -ring = 425/1500,0/3000 -congestion = 425/200,0/200,425/200,0/200,425/200,0/600 -callwaiting = 425/175,0/175,425/175,0/3500 -dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 -record = 1400/500,0/15000 -info = 950/330,0/1000 -dialout = 500 - - -[fi] -description = Finland -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -ringcadence = 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 -; -ringcadence = 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 -ringcadence = 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 diff --git a/1.2-netsec/configs/logger.conf.sample b/1.2-netsec/configs/logger.conf.sample deleted file mode 100644 index f2ff0ea7e..000000000 --- a/1.2-netsec/configs/logger.conf.sample +++ /dev/null @@ -1,69 +0,0 @@ -; -; 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 -; dtmf -; -; Special filename "console" represents the system console -; -; We highly recommend that you DO NOT turn on debug mode if you are simply -; running a production system. Debug mode turns on a LOT of extra messages, -; most of which you are unlikely to understand without an understanding of -; the underlying code. Do NOT report debug messages as code issues, unless -; you have a specific issue that you are attempting to debug. They are -; messages for just that -- debugging -- and do not rise to the level of -; something that merit your attention as an Asterisk administrator. Debug -; messages are also very verbose and can and do fill up logfiles quickly; -; this is another reason not to have debug mode on a production system unless -; you are in the process of debugging a specific issue. -; -;debug => debug -console => notice,warning,error -;console => notice,warning,error,debug -messages => notice,warning,error -;full => notice,warning,error,debug,verbose - -;syslog keyword : This special keyword logs to syslog facility -; -;syslog.local0 => notice,warning,error -; diff --git a/1.2-netsec/configs/manager.conf.sample b/1.2-netsec/configs/manager.conf.sample deleted file mode 100644 index ff37f8a1b..000000000 --- a/1.2-netsec/configs/manager.conf.sample +++ /dev/null @@ -1,37 +0,0 @@ -; -; 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 available 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 -; -; If the device connected via this user accepts input slowly, -; the timeout for writes to it can be increased to keep it -; from being disconnected (value is in milliseconds) -; -; writetimeout = 100 -; -; Authorization for various classes -;read = system,call,log,verbose,command,agent,user -;write = system,call,log,verbose,command,agent,user diff --git a/1.2-netsec/configs/meetme.conf.sample b/1.2-netsec/configs/meetme.conf.sample deleted file mode 100644 index 308ec0772..000000000 --- a/1.2-netsec/configs/meetme.conf.sample +++ /dev/null @@ -1,21 +0,0 @@ -; -; Configuration file for MeetMe simple conference rooms for Asterisk of course. -; -; This configuration file is read every time you call app meetme() - -[general] -;audiobuffers=32 ; The number of 20ms audio buffers to be used - ; when feeding audio frames from non-Zap channels - ; into the conference; larger numbers will allow - ; for the conference to 'de-jitter' audio that arrives - ; at different timing than the conference's timing - ; source, but can also allow for latency in hearing - ; the audio from the speaker. Minimum value is 2, - ; maximum value is 32. -; -[rooms] -; -; Usage is conf => confno[,pin][,adminpin] -; -;conf => 1234 -;conf => 2345,9938 diff --git a/1.2-netsec/configs/mgcp.conf.sample b/1.2-netsec/configs/mgcp.conf.sample deleted file mode 100644 index cf7b2c916..000000000 --- a/1.2-netsec/configs/mgcp.conf.sample +++ /dev/null @@ -1,75 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/misdn.conf.sample b/1.2-netsec/configs/misdn.conf.sample deleted file mode 100644 index 8957e2fa6..000000000 --- a/1.2-netsec/configs/misdn.conf.sample +++ /dev/null @@ -1,267 +0,0 @@ -; -; chan_misdn sample config -; - -; general section: -; -; for debugging and general setup, things that are not bound to port groups -; - -[general] - -; 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 - -; the big trace -; -; default value: [not set] -; -;tracefile=/var/log/misdn.trace - -; single call trace files -; set to true if you want to have them -; they depend on debug level -; -; default values: trace_calls : false -; trace_dir : /var/log/ -; -trace_calls=false -trace_dir=/var/log/ - -; set to yes if you want mISDN_dsp to bridge the calls in HW -; -; default value: yes -; -bridging=yes - -; stops dialtone after getting first digit on nt Port -; -; default value: yes -; -stop_tone_after_first_digit=yes - -; wether to append overlapdialed Digits to Extension or not -; -; default value: yes -; -append_digits2exten=yes - -; set this to yes if you have jollys mISDN which sends correct L1 Infos -; -; default value: yes -; -l1_info_ok=yes - -; set this to yes if you want to clear the l3 in case the l2 deactivates -; some environments have a flickering l2 which causes this option to -; damage active calls .. highly experimental -; -; default value: no -; -clear_l3=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 - -;;; CRYPTION STUFF - -; Wether 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 secions 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 - -; 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 espacially in NL seem to need this set to yes, also in -; switzerland this seems to be important -; -; default value: no -; -te_choose_channel=no - -; dialplan options: -; -; 0 - unknown -; 1 - National -; 2 - International -; 4 - Subscriber -; -; This setting is used for outgoing calls -; -; default value: 0 -; -dialplan=0 - -; This is only for asterisk head and will result in only considering -; misdn.confs and misdn_set_opts callingpresentation informations if set to no. -; Otherwise asterisks callingpresentation overwrites misdn.confs settings. -; -; default value: yes -; -use_callingpres=yes - -; 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 - -; 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 -; -; deafult values: not set = 0 -; -;callgroup=1 -;pickupgroup=1 - -; Allows/Screens Callerid -; -; possible values: allowed,not_screened -; -; be aware, if you set to allowed you need to set a correct -; callerid in the dialplan or set it here in the misdn.conf -; Some Telcos don't care about wrong callerids, others do ! -; -; default value: allowed -; -;presentation=not_screened - -; 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 -; -; default value: yes -; -echotraining=yes - -[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] -; if you want to have pp Protocol on one nt Port, you need -; to add a ptp directly after the portnumber, you can still add -; more ports and multiple ptp adds in your config. -ports=3ptp - -[first_extern] -; again port defs -ports=4 -; again a context for incomming 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 seperated 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 diff --git a/1.2-netsec/configs/modem.conf.sample b/1.2-netsec/configs/modem.conf.sample deleted file mode 100644 index 4bcd58abb..000000000 --- a/1.2-netsec/configs/modem.conf.sample +++ /dev/null @@ -1,92 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/modules.conf.sample b/1.2-netsec/configs/modules.conf.sample deleted file mode 100644 index 418433688..000000000 --- a/1.2-netsec/configs/modules.conf.sample +++ /dev/null @@ -1,53 +0,0 @@ -; -; Asterisk configuration file -; -; Module Loader configuration file -; - -[modules] -autoload=yes -; -; Any modules that need to be loaded before the Asterisk core has been -; initialized (just after the logger has been initialized) can be loaded -; using 'preload'. This will frequently be needed if you wish to map all -; module configuration files into Realtime storage, since the Realtime -; driver will need to be loaded before the modules using those configuration -; files are initialized. -; -; An example of loading ODBC support would be: -;preload => res_odbc.so -;preload => res_config_odbc.so -; -; If you want, load the GTK console right away. -; Don't load the KDE console since -; it's not as sophisticated right now. -; -noload => pbx_gtkconsole.so -;load => pbx_gtkconsole.so -noload => pbx_kdeconsole.so -; -; Intercom application is obsoleted by -; chan_oss. Don't load it. -; -noload => app_intercom.so -; -; The 'modem' channel driver and its subdrivers are -; obsolete, don't load them. -; -noload => chan_modem.so -noload => chan_modem_aopen.so -noload => chan_modem_bestdata.so -noload => chan_modem_i4l.so -; -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] diff --git a/1.2-netsec/configs/musiconhold.conf.sample b/1.2-netsec/configs/musiconhold.conf.sample deleted file mode 100644 index 6b3e7b694..000000000 --- a/1.2-netsec/configs/musiconhold.conf.sample +++ /dev/null @@ -1,64 +0,0 @@ -; -; Music on Hold -- Sample Configuration -; - -[default] -mode=quietmp3 -directory=/var/lib/asterisk/mohmp3 - -; valid mode options: -; quietmp3 -- default -; mp3 -- loud -; mp3nb -- unbuffered -; quietmp3nb -- quiet unbuffered -; custom -- run a custom application -; files -- read files from a directory in any Asterisk supported format - -;[manual] -;mode=custom -; Note that with mode=custom, a directory is not required, such as when reading -; from a stream. -;directory=/var/lib/asterisk/mohmp3 -;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s - -;[ulawstream] -;mode=custom -;application=/usr/bin/streamplayer 192.168.100.52 888 -;format=ulaw - -; mpg123 on Solaris does not always exit properly; madplay may be a better -; choice -;[solaris] -;mode=custom -;directory=/var/lib/asterisk/mohmp3 -;application=/site/sw/bin/madplay -Q -o raw:- --mono -R 8000 -a -12 -; - -; -; 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. -; - -;[native] -;mode=files -;directory=/var/lib/asterisk/moh-native -; -;[native-random] -;mode=files -;directory=/var/lib/asterisk/moh-native -;random=yes ; Play the files in a random order diff --git a/1.2-netsec/configs/osp.conf.sample b/1.2-netsec/configs/osp.conf.sample deleted file mode 100644 index e7e04036a..000000000 --- a/1.2-netsec/configs/osp.conf.sample +++ /dev/null @@ -1,64 +0,0 @@ -; -; 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 -; -; Defines the token format that Asterisk can validate. -; 0 - signed tokens only -; 1 - unsigned tokens only -; 2 - both signed and unsigned -; The defaults to 0, i.e. the Asterisk can validate signed tokens only. -; -;tokenformat=0 - -;[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 diff --git a/1.2-netsec/configs/oss.conf.sample b/1.2-netsec/configs/oss.conf.sample deleted file mode 100644 index 148a2a656..000000000 --- a/1.2-netsec/configs/oss.conf.sample +++ /dev/null @@ -1,39 +0,0 @@ -; -; 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 -; -; Set overridecontext to yes if you want the context specified above -; to override what someone places on the command line. -; -;overridecontext=yes -; -; 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 diff --git a/1.2-netsec/configs/phone.conf.sample b/1.2-netsec/configs/phone.conf.sample deleted file mode 100644 index ca028f9a1..000000000 --- a/1.2-netsec/configs/phone.conf.sample +++ /dev/null @@ -1,47 +0,0 @@ -; -; 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 diff --git a/1.2-netsec/configs/privacy.conf.sample b/1.2-netsec/configs/privacy.conf.sample deleted file mode 100644 index 0236bccb7..000000000 --- a/1.2-netsec/configs/privacy.conf.sample +++ /dev/null @@ -1,3 +0,0 @@ -[general] - -maxretries = 2 ;How many chances the caller has to enter their number diff --git a/1.2-netsec/configs/queues.conf.sample b/1.2-netsec/configs/queues.conf.sample deleted file mode 100644 index ba7a082b5..000000000 --- a/1.2-netsec/configs/queues.conf.sample +++ /dev/null @@ -1,200 +0,0 @@ -[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 -; an 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 answered 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 -; -; -; How often to make any periodic announcement (see periodic-announce) -; -;periodic-announce-frequency=60 -; -; 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") -;periodic-announce = queue-periodic-announce ; ("All reps busy / wait for next") -; -; 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 MONITOR_FILENAME=${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 - diff --git a/1.2-netsec/configs/res_odbc.conf.sample b/1.2-netsec/configs/res_odbc.conf.sample deleted file mode 100644 index 59d5c68c3..000000000 --- a/1.2-netsec/configs/res_odbc.conf.sample +++ /dev/null @@ -1,31 +0,0 @@ -;;; odbc setup file - -; ENV is a global set of environmental variables that will get set. -; Note that all environmental variables can be seen by all connections, -; so you can't have different values for different connections. -[ENV] -INFORMIXSERVER => my_special_database -INFORMIXDIR => /opt/informix - -; All other sections are arbitrary names for database connections. - -[asterisk] -enabled => yes -dsn => asterisk -;username => myuser -;password => mypass -pre-connect => yes - - -[mysql2] -enabled => no -dsn => MySQL-asterisk -username => myuser -password => mypass -pre-connect => yes - - - - - - diff --git a/1.2-netsec/configs/rpt.conf.sample b/1.2-netsec/configs/rpt.conf.sample deleted file mode 100644 index a66e50b92..000000000 --- a/1.2-netsec/configs/rpt.conf.sample +++ /dev/null @@ -1,180 +0,0 @@ -; Radio Repeater / Remote Base configuration file (for use with app_rpt) -; As of app_rpt version 0.36, 10/26/2005 -; - -;[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 -;;phone_functions = functions-phone ; (optional) different functions for 'P' mode -;;dphone_functions = functions-dphone ; (optional) different functions for 'D' mode -;;nodes = nodes-different ; (optional) different node list -;tonezone = us ; use US tones (default) -;context = default ; dialing context for phone -;callerid = "WB6NIL Repeater" <(213) 555-0123> ; 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 '#') -;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when - ; normal patch in use -;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 first 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 -;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 -;;phone_functions = functions-phone ; (optional) different functions for 'P' mode -;;dphone_functions = functions-dphone ; (optional) different functions for 'D' mode -;;nodes = nodes-different ; (optional) different node list -;tonezone = us ; use US tones (default) -;context = default ; dialing context for phone -;callerid = "WB6NIL Repeater" <(213) 555-0123> ; 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 '#') -;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when - ; normal patch in use -;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 = ft897 ; Set remote=y for dumb remote or - ; remote=ft897 for Yaesu FT-897 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@foo.bar.com/1234,foo.bar.com -;001 = context_B@baz.waldo.com/4321,baz.waldo.com -;002 = context_C@pepper.salt.com/5678,pepper.salt.com,y ; this is a remote - -;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 - diff --git a/1.2-netsec/configs/rtp.conf.sample b/1.2-netsec/configs/rtp.conf.sample deleted file mode 100644 index fa16f0d93..000000000 --- a/1.2-netsec/configs/rtp.conf.sample +++ /dev/null @@ -1,20 +0,0 @@ -; -; 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 -; -; The amount of time a DTMF digit with no 'end' marker should be -; allowed to continue (in 'samples', 1/8000 of a second) -; -;dtmftimeout=3000 diff --git a/1.2-netsec/configs/sip.conf.sample b/1.2-netsec/configs/sip.conf.sample deleted file mode 100644 index 3d9299c05..000000000 --- a/1.2-netsec/configs/sip.conf.sample +++ /dev/null @@ -1,441 +0,0 @@ -; -; SIP Configuration example for Asterisk -; -; Syntax for specifying a SIP device in extensions.conf is -; SIP/devicename where devicename is defined in a section below. -; -; You may also use -; SIP/username@domain to call any SIP user on the Internet -; (Don't forget to enable DNS SRV records if you want to use this) -; -; If you define a SIP proxy as a peer below, you may call -; SIP/proxyhostname/user or SIP/user@proxyhostname -; where the proxyhostname is defined in a section below -; -; Useful CLI commands to check peers/users: -; sip show peers Show all SIP peers (including friends) -; sip show users Show all SIP users (including friends) -; sip show registry Show status of hosts we register with -; -; sip debug Show all SIP messages -; -; reload chan_sip.so Reload configuration file -; Active SIP peers will not be reconfigured -; - -[general] -context=default ; Default context for incoming calls -;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' - ; if asterisk was compiled with OSP support. -;realm=mydomain.tld ; Realm for digest authentication - ; defaults to "asterisk" - ; Realms MUST be globally unique according to RFC 3261 - ; Set this to your host name or domain name -bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) -bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) -srvlookup=yes ; Enable DNS SRV lookups on outbound calls - ; Note: Asterisk only uses the first host - ; in SRV records - ; Disabling DNS SRV lookups disables the - ; ability to place SIP calls based on domain - ; names to some other SIP users on the Internet - -;domain=mydomain.tld ; Set default domain for this host - ; If configured, Asterisk will only allow - ; INVITE and REFER to non-local domains - ; Use "sip show domains" to list local domains -;domain=mydomain.tld,mydomain-incoming - ; Add domain and configure incoming context - ; for external calls to this domain -;domain=1.2.3.4 ; Add IP address as local domain - ; You can have several "domain" settings -;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains - ; Default is yes -;autodomain=yes ; Turn this on to have Asterisk add local host - ; name and local IP to domain list. -;pedantic=yes ; Enable slow, pedantic checking for Pingtel - ; and multiline formatted headers for strict - ; SIP compatibility (defaults to "no") -;tos=184 ; Set IP QoS to either a keyword or numeric val -;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none -;maxexpiry=3600 ; Max length of incoming registration we allow -;defaultexpiry=120 ; Default length of incoming/outoing registration -;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY -;checkmwi=10 ; Default time between mailbox checks for peers -;vmexten=voicemail ; dialplan extension to reach mailbox sets the - ; Message-Account in the MWI notify message - ; defaults to "asterisk" -;videosupport=yes ; Turn on support for SIP video -;recordhistory=yes ; Record SIP history by default - ; (see sip history / sip no history) - -;disallow=all ; First disallow all codecs -;allow=ulaw ; Allow codecs in order of preference -;allow=ilbc ; -;musicclass=default ; Sets the default music on hold class for all SIP calls - ; This may also be set for individual users/peers -;language=en ; Default language setting for all users/peers - ; This may also be set for individual users/peers -;relaxdtmf=yes ; Relax dtmf handling -;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity - ; when we're not on hold -;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity - ; when we're on hold (must be > rtptimeout) -;trustrpid = no ; If Remote-Party-ID should be trusted -;sendrpid = yes ; If Remote-Party-ID should be sent -;progressinband=never ; If we should generate in-band ringing always - ; use 'never' to never use in-band signalling, even in cases - ; where some buggy devices might not render it -;useragent=Asterisk PBX ; Allows you to change the user agent string -;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address - ; Note that promiscredir when redirects are made to the - ; local system will cause loops since SIP is incapable - ; of performing a "hairpin" call. -;usereqphone = no ; If yes, ";user=phone" is added to uri that contains - ; a valid phone number -;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 - ; Other options: - ; info : SIP INFO messages - ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) - ; auto : Use rfc2833 if offered, inband otherwise - -;compactheaders = yes ; send compact sip headers. -;sipdebug = yes ; Turn on SIP debugging by default, from - ; the moment the channel loads this configuration -;subscribecontext = default ; Set a specific context for SUBSCRIBE requests - ; Useful to limit subscriptions to local extensions - ; Settable per peer/user also -;notifyringing = yes ; Notify subscriptions on RINGING state - -; -; 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=sipregistrations -; -; Asterisk can register as a SIP user agent to a SIP proxy (provider) -; Format for the register statement is: -; register => user[:secret[:authuser]]@host[:port][/extension] -; -; If no extension is given, the 's' extension is used. The extension needs to -; be defined in extensions.conf to be able to accept calls from this SIP proxy -; (provider). -; -; host is either a host name defined in DNS or the name of a section defined -; below. -; -; Examples: -; -;register => 1234:password@mysipprovider.com -; -; This will pass incoming calls to the 's' extension -; -; -;register => 2345:password@sip_proxy/1234 -; -; Register 2345 at sip provider 'sip_proxy'. Calls from this provider -; connect to local extension 1234 in extensions.conf, default context, -; unless you configure a [sip_proxy] section below, and configure a -; context. -; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] -; Tip 2: Use separate type=peer and type=user sections for SIP providers -; (instead of type=friend) if you have calls in both directions - -;registertimeout=20 ; retry registration calls every 20 seconds (default) -;registerattempts=10 ; Number of registration attempts before we give up - ; 0 = continue forever, hammering the other server until it - ; accepts the registration - ; Default is 0 tries, continue forever -;callevents=no ; generate manager events when sip ua performs events (e.g. hold) - -;----------------------------------------- NAT SUPPORT ------------------------ -; The externip, externhost and localnet settings are used if you use Asterisk -; behind a NAT device to communicate with services on the outside. - -;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages - ; if we're behind a NAT - - ; The externip and localnet is used - ; when registering and communicating with other proxies - ; that we're registered with -;externhost=foo.dyndns.net ; Alternatively you can specify an - ; external host, and Asterisk will - ; perform DNS queries periodically. Not - ; recommended for production - ; environments! Use externip instead -;externrefresh=10 ; How often to refresh externhost if - ; used - ; You may add multiple local networks. A reasonable set of defaults - ; are: -;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks -;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 -;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation -;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network - -; The nat= setting is used when Asterisk is on a public IP, communicating with -; devices hidden behind a NAT device (broadband router). If you have one-way -; audio problems, you usually have problems with your NAT configuration or your -; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP -; ports for incoming audio in rtp.conf -; -;nat=no ; Global NAT settings (Affects all peers and users) - ; yes = Always ignore info and assume NAT - ; no = Use NAT mode only according to RFC3581 - ; never = Never attempt NAT mode or RFC3581 support - ; route = Assume NAT, don't send rport - ; (work around more UNIDEN bugs) - -;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list - ; just like friends added from the config file only on a - ; as-needed basis? (yes|no) - -;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) - ; If set to yes, when a SIP UA registers successfully, the ip address, - ; the origination port, the registration period, and the username of - ; the UA will be set to database via realtime. If not present, defaults to 'yes'. - -;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule - ; as if it had just registered? (yes|no|<seconds>) - ; If set to yes, 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 - ; registration interval. - -;ignoreregexpire=yes ; Enabling this setting has two functions: - ; - ; For non-realtime peers, when their registration expires, the information - ; will _not_ be removed from memory or the Asterisk database; if you attempt - ; to place a call to the peer, the existing information will be used in spite - ; of it having expired - ; - ; For realtime peers, when the peer is retrieved from realtime storage, - ; the registration information will be used regardless of whether - ; it has expired or not; if it expires while the realtime peer is still in - ; memory (due to caching or other reasons), the information will not be - ; removed from realtime storage - -; Incoming INVITE and REFER messages can be matched against a list of 'allowed' -; domains, each of which can direct the call to a specific context if desired. -; By default, all domains are accepted and sent to the default context or the -; context associated with the user/peer placing the call. -; Domains can be specified using: -; domain=<domain>[,<context>] -; Examples: -; domain=myasterisk.dom -; domain=customer.com,customer-context -; -; In addition, all the 'default' domains associated with a server should be -; added if incoming request filtering is desired. -; autodomain=yes -; -; To disallow requests for domains not serviced by this server: -; allowexternaldomains=no - -; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to - ; non-peers, use your primary domain "identity" - ; for From: headers instead of just your IP - ; address. This is to be polite and - ; it may be a mandatory requirement for some - ; destinations which do not have a prior - ; account relationship with your server. - -[authentication] -; Global credentials for outbound calls, i.e. when a proxy challenges your -; Asterisk server for authentication. These credentials override -; any credentials in peer/register definition if realm is matched. -; -; This way, Asterisk can authenticate for outbound calls to other -; realms. We match realm on the proxy challenge and pick an set of -; credentials from this list -; Syntax: -; auth = <user>:<secret>@<realm> -; auth = <user>#<md5secret>@<realm> -; Example: -;auth=mark:topsecret@digium.com -; -; You may also add auth= statements to [peer] definitions -; Peer auth= override all other authentication settings if we match on realm - -;------------------------------------------------------------------------------ -; Users and peers have different settings available. Friends have all settings, -; since a friend is both a peer and a user -; -; User config options: Peer configuration: -; -------------------- ------------------- -; context context -; permit permit -; deny deny -; secret secret -; md5secret md5secret -; dtmfmode dtmfmode -; canreinvite canreinvite -; nat nat -; callgroup callgroup -; pickupgroup pickupgroup -; language language -; allow allow -; disallow disallow -; insecure insecure -; trustrpid trustrpid -; progressinband progressinband -; promiscredir promiscredir -; useclientcode useclientcode -; accountcode accountcode -; setvar setvar -; callerid callerid -; amaflags amaflags -; call-limit call-limit -; restrictcid restrictcid -; subscribecontext subscribecontext -; mailbox -; username -; template -; fromdomain -; regexten -; fromuser -; host -; port -; qualify -; defaultip -; rtptimeout -; rtpholdtimeout -; sendrpid - -;[sip_proxy] -; For incoming calls only. Example: FWD (Free World Dialup) -; We match on IP address of the proxy for incoming calls -; since we can not match on username (caller id) -;type=peer -;context=from-fwd -;host=fwd.pulver.com - -;[sip_proxy-out] -;type=peer ; we only want to call out, not be called -;secret=guessit -;username=yourusername ; Authentication user for outbound proxies -;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain -;host=box.provider.com -;usereqphone=yes ; This provider requires ";user=phone" on URI -;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer - -;------------------------------------------------------------------------------ -; Definitions of locally connected SIP phones -; -; type = user a device that authenticates to us by "from" field to place calls -; type = peer a device we place calls to or that calls us and we match by host -; type = friend two configurations (peer+user) in one -; -; For local phones, type=friend works most of the time -; -; If you have one-way audio, you propably have NAT problems. -; If Asterisk is on a public IP, and the phone is inside of a NAT device -; you will need to configure nat option for those phones. -; Also, turn on qualify=yes to keep the nat session open - -;[grandstream1] -;type=friend -;context=from-sip ; Where to start in the dialplan when this phone calls -;callerid=John Doe <1234> ; Full caller ID, to override the phones config -;host=192.168.0.23 ; we have a static but private IP address - ; No registration allowed -;nat=no ; there is not NAT between phone and Asterisk -;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk -;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone -;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time - ; from the phone to asterisk - ; (1 for the explicit peer, 1 for the explicit user, - ; remember that a friend equals 1 peer and 1 user in - ; memory) -;mailbox=1234@default ; mailbox 1234 in voicemail context "default" -;disallow=all ; need to disallow=all before we can use allow= -;allow=ulaw ; Note: In user sections the order of codecs - ; listed with allow= does NOT matter! -;allow=alaw -;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! -;allow=g729 ; Pass-thru only unless g729 license obtained -;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists - - -;[xlite1] -; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! -; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed -;type=friend -;regexten=1234 ; When they register, create extension 1234 -;callerid="Jane Smith" <5678> -;host=dynamic ; This device needs to register -;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 -;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes - - -;[snom] -;type=friend ; Friends place calls and receive calls -;context=from-sip ; Context for incoming calls from this user -;secret=blah -;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions -;language=de ; Use German prompts for this user -;host=dynamic ; This peer register with us -;dtmfmode=inband ; Choices are inband, rfc2833, or info -;defaultip=192.168.0.59 ; IP used until peer registers -;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator -;vmexten=voicemail ; dialplan extension to reach mailbox - ; sets the Message-Account in the MWI notify message - ; defaults to global vmexten which defaults to "asterisk" -;restrictcid=yes ; To have the callerid restriced -> sent as ANI -;disallow=all -;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! - - -;[polycom] -;type=friend ; Friends place calls and receive calls -;context=from-sip ; Context for incoming calls from this user -;secret=blahpoly -;host=dynamic ; This peer register with us -;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info -;username=polly ; Username to use in INVITE until peer registers - ; Normally you do NOT need to set this parameter -;disallow=all -;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! -;progressinband=no ; Polycom phones don't work properly with "never" - - -;[pingtel] -;type=friend -;secret=blah -;host=dynamic -;insecure=port ; Allow matching of peer by IP address without matching port number -;insecure=invite ; Do not require authentication of incoming INVITEs -;insecure=port,invite ; (both) -;qualify=1000 ; Consider it down if it's 1 second to reply - ; Helps with NAT session - ; qualify=yes uses default value -;callgroup=1,3-4 ; We are in caller groups 1,3,4 -;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 -;defaultip=192.168.0.60 ; IP address to use if peer has not registred - -;[cisco1] -;type=friend -;secret=blah -;qualify=200 ; Qualify peer is no more than 200ms away -;nat=yes ; This phone may be natted - ; Send SIP and RTP to the IP address that packet is - ; received from instead of trusting SIP headers -;host=dynamic ; This device registers with us -;canreinvite=no ; Asterisk by default tries to redirect the - ; RTP media stream (audio) to go directly from - ; the caller to the callee. Some devices do not - ; support this (especially if one of them is - ; behind a NAT). -;defaultip=192.168.0.4 ; IP address to use until registration -;username=goran ; Username to use when calling this device before registration - ; Normally you do NOT need to set this parameter -;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device - diff --git a/1.2-netsec/configs/sip_notify.conf.sample b/1.2-netsec/configs/sip_notify.conf.sample deleted file mode 100644 index 8b10da555..000000000 --- a/1.2-netsec/configs/sip_notify.conf.sample +++ /dev/null @@ -1,22 +0,0 @@ -[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 diff --git a/1.2-netsec/configs/skinny.conf.sample b/1.2-netsec/configs/skinny.conf.sample deleted file mode 100644 index d57923d85..000000000 --- a/1.2-netsec/configs/skinny.conf.sample +++ /dev/null @@ -1,55 +0,0 @@ -; -; 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 - - diff --git a/1.2-netsec/configs/telcordia-1.adsi b/1.2-netsec/configs/telcordia-1.adsi deleted file mode 100644 index 1486aa95e..000000000 --- a/1.2-netsec/configs/telcordia-1.adsi +++ /dev/null @@ -1,83 +0,0 @@ -; -; Asterisk default ADSI script -; -; -; Begin with the preamble requirements -; -DESCRIPTION "Telcordia Demo" ; Name of vendor -VERSION 0x02 ; Version of stuff -;SECURITY "_AST" ; Security code -SECURITY 0x0000 ; Security code -FDN 0x0000000f ; Descriptor number - -; -; Predefined strings -; -DISPLAY "talkingto" IS "Talking To" "$Call1p" WRAP -DISPLAY "titles" IS "20th Century IQ Svc" -DISPLAY "newcall" IS "New Call From" "$Call1p" WRAP -DISPLAY "ringing" IS "Ringing" - -; -; Begin state definitions -; -STATE "callup" ; Call is currently up -STATE "inactive" ; No active call - -; -; Begin soft key definitions -; -KEY "CB_OH" IS "Block" OR "Call Block" - OFFHOOK - VOICEMODE - WAITDIALTONE - SENDDTMF "*60" - SUBSCRIPT "offHook" -ENDKEY - -KEY "CB" IS "Block" OR "Call Block" - SENDDTMF "*60" -ENDKEY - -; -; Begin main subroutine -; - -SUB "main" IS - IFEVENT NEARANSWER THEN - CLEAR - SHOWDISPLAY "talkingto" AT 1 - GOTO "stableCall" - ENDIF - IFEVENT OFFHOOK THEN - CLEAR - SHOWDISPLAY "titles" AT 1 - SHOWKEYS "CB" - GOTO "offHook" - ENDIF - IFEVENT IDLE THEN - CLEAR - SHOWDISPLAY "titles" AT 1 - SHOWKEYS "CB_OH" - ENDIF - IFEVENT CALLERID THEN - CLEAR - SHOWDISPLAY "newcall" AT 1 - ENDIF -ENDSUB - -SUB "offHook" IS - IFEVENT FARRING THEN - CLEAR - SHOWDISPLAY "ringing" AT 1 - ENDIF - IFEVENT FARANSWER THEN - CLEAR - SHOWDISPLAY "talkingto" AT 1 - GOTO "stableCall" - ENDIF -ENDSUB - -SUB "stableCall" IS - -ENDSUB diff --git a/1.2-netsec/configs/voicemail.conf.sample b/1.2-netsec/configs/voicemail.conf.sample deleted file mode 100644 index fa362c95d..000000000 --- a/1.2-netsec/configs/voicemail.conf.sample +++ /dev/null @@ -1,213 +0,0 @@ -; -; Voicemail Configuration -; - -; -; NOTE: Asterisk has to edit this file to change a user's password. This does -; note currently work with the "#include <file>" directive for Asterisk -; configuration files. Do not use it with this configuration file. -; - -[general] -; Default formats for writing Voicemail -;format=g723sf|wav49|wav -format=wav49|gsm|wav -; -; WARNING: -; If you change the list of formats that you record voicemail in -; when you have mailboxes that contain messages, you _MUST_ absolutely -; manually go through those mailboxes and convert/delete/add the -; the message files so that they appear to have been stored using -; your new format list. If you don't do this, very unpleasant -; things may happen to your users while they are retrieving and -; manipulating their voicemail. -; -; In other words: don't change the format list on a production system -; unless you are _VERY_ sure that you know what you are doing and are -; prepared for the consequences. -; -; Who the e-mail notification should appear to come from -serveremail=asterisk -;serveremail=asterisk@linux-support.net -; Should the email contain the voicemail as an attachment -attach=yes -; Maximum number of messages per folder. If not specified, a default value -; (100) is used. Maximum value for this option is 9999. -;maxmsg=100 -; Maximum length of a voicemail message in seconds -;maxmessage=180 -; Minimum length of a voicemail message in seconds for the message to be kept -; The default is no minimum. -;minmessage=3 -; Maximum length of greetings in seconds -;maxgreet=60 -; How many miliseconds to skip forward/back when rew/ff in message playback -skipms=3000 -; How many seconds of silence before we end the recording -maxsilence=10 -; Silence threshold (what we consider silence, the lower, the more sensitive) -silencethreshold=128 -; Max number of failed login attempts -maxlogins=3 -; 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 -; For the directory, you can override the intro file if you want -;directoryintro=dir-intro -; The character set for voicemail messages can be specified here -;charset=ISO-8859-1 -; The ADSI feature descriptor number to download to -;adsifdn=0000000F -; The ADSI security lock code -;adsisec=9BDBF7AC -; The ADSI voicemail application version number. -;adsiver=1 -; Skip the "[PBX]:" string from the message title -;pbxskip=yes -; Change the From: string -;fromstring=The Asterisk PBX -; Permit finding entries for forward/compose from the directory -;usedirectory=yes -; -; Change the from, 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 only be up to 512 characters due to a -; limitation in the Asterisk configuration subsystem. -;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, otherise 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 also change the Pager From: string, the pager body and/or subject. -; The above defined variables also can be used here -;pagerfromstring=The Asterisk PBX -;pagersubject=New VM -;pagerbody=New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE} -; -; Set the date format on outgoing mails. Valid arguments can be found on the -; strftime(3) man page -; -; Default -emaildateformat=%A, %B %d, %Y at %r -; 24h date format -;emaildateformat=%A, %d %B %Y at %H:%M:%S -; -; You can override the default program to send e-mail if you wish, too -; -;mailcmd=/usr/sbin/sendmail -t -; -; Users may be located in different timezones, or may have different -; message announcements for their introductory message when they enter -; the voicemail system. Set the message and the timezone each user -; hears here. Set the user into one of these zones with the tz= attribute -; in the options field of the mailbox. Of course, language substitution -; still applies here so you may have several directory trees that have -; alternate language choices. -; -; Look in /usr/share/zoneinfo/ for names of timezones. -; Look at the manual page for strftime for a quick tutorial on how the -; variable substitution is done on the values below. -; -; Supported values: -; 'filename' filename of a soundfile (single ticks around the filename -; required) -; ${VAR} variable substitution -; A or a Day of week (Saturday, Sunday, ...) -; B or b or h Month name (January, February, ...) -; d or e numeric day of month (first, second, ..., thirty-first) -; Y Year -; I or l Hour, 12 hour clock -; H Hour, 24 hour clock (single digit hours preceded by "oh") -; k Hour, 24 hour clock (single digit hours NOT preceded by "oh") -; M Minute, with 00 pronounced as "o'clock" -; N Minute, with 00 pronounced as "hundred" (US military time) -; P or p AM or PM -; Q "today", "yesterday" or ABdY -; (*note: not standard strftime value) -; q "" (for today), "yesterday", weekday, or ABdY -; (*note: not standard strftime value) -; R 24 hour time, including minute -; -; - -; -; Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options> -; if the e-mail is specified, a message will be sent when a message is -; received, to the given mailbox. If pager is specified, a message will be -; sent there as well. If the password is prefixed by '-', then it is -; considered to be unchangable. -; -; 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 -; searchcontexts=yes ; Current default behavior is to search only the default context - ; if one is not specified. The older behavior was to search all contexts. - ; This option restores the old behavior [DEFAULT=no] -; 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. - ; Note: deletevoicemail is provided as an equivalent option for Realtime configuration. -; 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 N 'hours' -military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' - -[default] -; Define maximum number of messages per folder for partcular context. -;maxmsg=50 - -1234 => 4242,Example Mailbox,root@localhost -;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central|maxmsg=10 -;4300 => 3456,Ben Rigas,ben@american-computer.net -;4310 => -5432,Sales,sales@marko.net -;4069 => 6522,Matt Brooks,matt@marko.net,,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1 -;4073 => 1099,Bianca Paige,bianca@biancapaige.com,,delete=1 -;4110 => 3443,Rob Flynn,rflynn@blueridge.net - - -; -; Mailboxes may be organized into multiple contexts for -; voicemail virtualhosting -; - -[other] -;The intro can be customized on a per-context basis -;directoryintro=dir-company2 -1234 => 5678,Company2 User,root@localhost diff --git a/1.2-netsec/configs/vpb.conf.sample b/1.2-netsec/configs/vpb.conf.sample deleted file mode 100644 index d16283802..000000000 --- a/1.2-netsec/configs/vpb.conf.sample +++ /dev/null @@ -1,108 +0,0 @@ -; -; V6PCI/V12PCI config file for VoiceTronix Hardware -; -; Options for [general] section -; -; type = v12pci|v6pci|v4pci -; cards = number of cards -; To use Asterisk indication tones -; indication = 1 -; none,-24db,-18db only for use with OpenLine4 -; ecsuppthres = 0|2048|4096 -; Inter Digit Delay timeout for when collecting DTMF tones for dialling -; from a Station port, in ms -; dtmfidd = 3000 -; To use Asterisk DTMF detection -; ast-dtmf-det=1 -; Used with ast-dtmf-det -; relaxdtmf=1 -; When a native bridge occurs between 2 vpb channels, it will only break -; the connection for '#' and '*' -; break-for-dtmf=no -; Set the maximum period between received rings, default 4000ms -; timer_period_ring=4000 -; -; Options 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 - - diff --git a/1.2-netsec/configs/zapata.conf.sample b/1.2-netsec/configs/zapata.conf.sample deleted file mode 100644 index 06aa48283..000000000 --- a/1.2-netsec/configs/zapata.conf.sample +++ /dev/null @@ -1,569 +0,0 @@ -; -; Zapata telephony interface -; -; Configuration file -; -; You need to restart Asterisk to re-configure the Zap channel -; CLI> reload chan_zap.so -; will reload the configuration file, -; but not all configuration options are -; re-configured during a reload. - - - -[trunkgroups] -; -; Trunk groups are used for NFAS or GR-303 connections. -; -; Group: Defines a trunk group. -; group => <trunkgroup>,<dchannel>[,<backup1>...] -; -; trunkgroup is the numerical trunk group to create -; dchannel is the zap channel which will have the -; d-channel for the trunk. -; backup1 is an optional list of backup d-channels. -; -;trunkgroup => 1,24,48 -;trunkgroup => 1,24 -; -; Spanmap: Associates a span with a trunk group -; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>] -; -; zapspan is the zap span number to associate -; trunkgroup is the trunkgroup (specified above) for the mapping -; logicalspan is the logical span number within the trunk group to use. -; if unspecified, no logical span number is used. -; -;spanmap => 1,1,1 -;spanmap => 2,1,2 -;spanmap => 3,1,3 -;spanmap => 4,1,4 - -[channels] -; -; Default language -; -;language=en -; -; Default context -; -context=default -; -; Switchtype: Only used for PRI. -; -; national: National ISDN 2 (default) -; dms100: Nortel DMS100 -; 4ess: AT&T 4ESS -; 5ess: Lucent 5ESS -; euroisdn: EuroISDN -; ni1: Old National ISDN 1 -; qsig: Q.SIG -; -switchtype=national -; -; Some switches (AT&T especially) require network specific facility IE -; supported values are currently 'none', 'sdn', 'megacom', 'accunet' -; -;nsf=none -; -; PRI Dialplan: Only RARELY used for PRI. -; -; unknown: Unknown -; private: Private ISDN -; local: Local ISDN -; national: National ISDN -; international: International ISDN -; -;pridialplan=national -; -; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) -; -; unknown: Unknown -; private: Private ISDN -; local: Local ISDN -; national: National ISDN -; international: International ISDN -; -;prilocaldialplan=national -; -; PRI callerid prefixes based on the given TON/NPI (dialplan) -; This is especially needed for euroisdn E1-PRIs -; -; sample 1 for Germany -;internationalprefix = 00 -;nationalprefix = 0 -;localprefix = 0711 -;privateprefix = 07115678 -;unknownprefix = -; -; sample 2 for Germany -;internationalprefix = + -;nationalprefix = +49 -;localprefix = +49711 -;privateprefix = +497115678 -;unknownprefix = -; -; PRI resetinterval: sets the time in seconds between restart of unused -; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like -; channel restarts. so set the interval to a very long interval e.g. 100000000 -; or 'never' to disable *entirely*. -; -;resetinterval = 3600 -; -; Overlap dialing mode (sending overlap digits) -; -;overlapdial=yes -; -; PRI Out of band indications. -; Enable this to report Busy and Congestion on a PRI using out-of-band -; notification. Inband indication, as used by Asterisk doesn't seem to work -; with all telcos. -; -; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT -; inband: Signal Busy/Congestion using in-band tones -; -; priindication = outofband -; -; If you need to override the existing channels selection routine and force all -; PRI channels to be marked as exclusively selected, set this to yes. -; priexclusive = yes -; -; ISDN Timers -; All of the ISDN timers and counters that are used are configurable. Specify -; the timer name, and its value (in ms for timers). -; -; pritimer => t200,1000 -; pritimer => t313,4000 -; -; To enable transmission of facility-based ISDN supplementary services (such -; as caller name from CPE over facility), enable this option. -; facilityenable = yes -; -; -; Signalling method (default is fxs). Valid values: -; em: E & M -; em_w: E & M Wink -; featd: Feature Group D (The fake, Adtran style, DTMF) -; featdmf: Feature Group D (The real thing, MF (domestic, US)) -; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through -; a Tandem Access point -; featb: Feature Group B (MF (domestic, US)) -; fxs_ls: FXS (Loop Start) -; fxs_gs: FXS (Ground Start) -; fxs_ks: FXS (Kewl Start) -; fxo_ls: FXO (Loop Start) -; fxo_gs: FXO (Ground Start) -; fxo_ks: FXO (Kewl Start) -; pri_cpe: PRI signalling, CPE side -; pri_net: PRI signalling, Network side -; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side -; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side -; sf: SF (Inband Tone) Signalling -; sf_w: SF Wink -; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) -; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) -; sf_featb: SF Feature Group B (MF (domestic, US)) -; e911: E911 (MF) style signalling -; -; The following are used for Radio interfaces: -; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the -; channel bank) -; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the -; channel bank) -; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the -; channel bank) -; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at -; the channel bank) -; em_rx: Receive audio/COR on an E&M interface (1-way) -; em_tx: Transmit audio/PTT on an E&M interface (1-way) -; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface -; (2-way) -; em_rxtx: Same as em_txrx (for our dyslexic friends) -; sf_rx: Receive audio/COR on an SF interface (1-way) -; sf_tx: Transmit audio/PTT on an SF interface (1-way) -; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface -; (2-way) -; sf_rxtx: Same as sf_txrx (for our dyslexic friends) -; -signalling=fxo_ls -; -; For Feature Group D Tandem access, to set the default CIC and OZZ use these -; parameters: -;defaultozz=0000 -;defaultcic=303 -; -; A variety of timing parameters can be specified as well -; Including: -; prewink: Pre-wink time (default 50ms) -; preflash: Pre-flash time (default 50ms) -; wink: Wink time (default 150ms) -; flash: Flash time (default 750ms) -; start: Start time (default 1500ms) -; rxwink: Receiver wink time (default 300ms) -; rxflash: Receiver flashtime (default 1250ms) -; debounce: Debounce timing (default 600ms) -; -rxwink=300 ; Atlas seems to use long (250ms) winks -; -; How long generated tones (DTMF and MF) will be played on the channel -; (in miliseconds) -;toneduration=100 -; -; Whether or not to do distinctive ring detection on FXO lines -; -;usedistinctiveringdetection=yes - -; -; Whether or not to use caller ID -; -usecallerid=yes -; -; Type of caller ID signalling in use -; bell = bell202 as used in US -; v23 = v23 as used in the UK -; dtmf = DTMF as used in Denmark, Sweden and Netherlands -; -;cidsignalling=bell -; -; What signals the start of caller ID -; ring = a ring signals the start -; polarity = polarity reversal signals the start -; -;cidstart=ring -; -; Whether or not to hide outgoing caller ID (Override with *67 or *82) -; -hidecallerid=no -; -; Whether or not to enable call waiting on FXO lines -; -callwaiting=yes -; -; Whether or not restrict outgoing caller ID (will be sent as ANI only, not -; available for the user) -; Mostly use with FXS ports -; -;restrictcid=no -; -; Whether or not use the caller ID presentation for the outgoing call that the -; calling switch is sending. -; -usecallingpres=yes -; -; Some countries (UK) have ring tones with different ring tones (ring-ring), -; which means the callerid needs to be set later on, and not just after -; the first ring, as per the default. -; -;sendcalleridafter=1 -; -; -; Support Caller*ID on Call Waiting -; -callwaitingcallerid=yes -; -; Support three-way calling -; -threewaycalling=yes -; -; Support flash-hook call transfer (requires three way calling) -; Also enables call parking (overrides the 'canpark' parameter) -; -transfer=yes -; -; Allow call parking -; ('canpark=no' is overridden by 'transfer=yes') -; -canpark=yes -; -; Support call forward variable -; -cancallforward=yes -; -; Whether or not to support Call Return (*69) -; -callreturn=yes -; -; Stutter dialtone support: If a mailbox is specified without a voicemail -; context, then when voicemail is received in a mailbox in the default -; voicemail context in voicemail.conf, taking the phone off hook will cause a -; stutter dialtone instead of a normal one. -; -; If a mailbox is specified *with* a voicemail context, the same will result -; if voicemail recieved in mailbox in the specified voicemail context. -; -; for default voicemail context, the example below is fine: -; -;mailbox=1234 -; -; for any other voicemail context, the following will produce the stutter tone: -; -;mailbox=1234@context -; -; Enable echo cancellation -; Use either "yes", "no", or a power of two from 32 to 256 if you wish to -; actually set the number of taps of cancellation. -; -echocancel=yes -; -; Generally, it is not necessary (and in fact undesirable) to echo cancel when -; the circuit path is entirely TDM. You may, however, reverse this behavior -; by enabling the echo cancel during pure TDM bridging below. -; -echocancelwhenbridged=yes -; -; In some cases, the echo canceller doesn't train quickly enough and there -; is echo at the beginning of the call. Enabling echo training will cause -; asterisk to briefly mute the channel, send an impulse, and use the impulse -; response to pre-train the echo canceller so it can start out with a much -; closer idea of the actual echo. Value may be "yes", "no", or a number of -; milliseconds to delay before training (default = 400) -; -;echotraining=yes -;echotraining=800 -; -; If you are having trouble with DTMF detection, you can relax the DTMF -; detection parameters. Relaxing them may make the DTMF detector more likely -; to have "talkoff" where DTMF is detected when it shouldn't be. -; -;relaxdtmf=yes -; -; You may also set the default receive and transmit gains (in dB) -; -rxgain=0.0 -txgain=0.0 -; -; Logical groups can be assigned to allow outgoing rollover. Groups range -; from 0 to 63, and multiple groups can be specified. -; -group=1 -; -; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing -; and it is a member of a group which is one of your pickup groups, then -; you can answer it by picking up and dialing *8#. For simple offices, just -; make these both the same -; -callgroup=1 -pickupgroup=1 - -; -; Specify whether the channel should be answered immediately or if the simple -; switch should provide dialtone, read digits, etc. -; -immediate=no -; -; Specify whether flash-hook transfers to 'busy' channels should complete or -; return to the caller performing the transfer (default is yes). -; -;transfertobusy=no -; -; CallerID can be set to "asreceived" or a specific number if you want to -; override it. Note that "asreceived" only applies to trunk interfaces. -; -;callerid=2564286000 -; -; AMA flags affects the recording of Call Detail Records. If specified -; it may be 'default', 'omit', 'billing', or 'documentation'. -; -;amaflags=default -; -; Channels may be associated with an account code to ease -; billing -; -;accountcode=lss0101 -; -; ADSI (Analog Display Services Interface) can be enabled on a per-channel -; basis if you have (or may have) ADSI compatible CPE equipment -; -;adsi=yes -; -; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D -; etc, it can be useful to perform busy detection either in an effort to -; detect hangup or for detecting busies. This enables listening for -; the beep-beep busy pattern. -; -;busydetect=yes -; -; If busydetect is enabled, it is also possible to specify how many busy tones -; to wait for before hanging up. The default is 4, but better results can be -; achieved if set to 6 or even 8. Mind that the higher the number, the more -; time that will be needed to hangup a channel, but lowers the probability -; that you will get random hangups. -; -;busycount=4 -; -; If busydetect is enabled, it is also possible to specify the cadence of your -; busy signal. In many countries, it is 500msec on, 500msec off. Without -; busypattern specified, we'll accept any regular sound-silence pattern that -; repeats <busycount> times as a busy signal. If you specify busypattern, -; then we'll further check the length of the sound (tone) and silence, which -; will further reduce the chance of a false positive. -; -;busypattern=500,500 -; -; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy -; detector. If your country has a busy tone with the same length tone and -; silence (as many countries do), consider defining the -; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option. -; -; Use a polarity reversal to mark when a outgoing call is answered by the -; remote party. -; -;answeronpolarityswitch=yes -; -; In some countries, a polarity reversal is used to signal the disconnect of a -; phone line. If the hanguponpolarityswitch option is selected, the call will -; be considered "hung up" on a polarity reversal. -; -;hanguponpolarityswitch=yes -; -; On trunk interfaces (FXS) it can be useful to attempt to follow the progress -; of a call through RINGING, BUSY, and ANSWERING. If turned on, call -; progress attempts to determine answer, busy, and ringing on phone lines. -; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, -; so don't count on it being very accurate. -; -; Few zones are supported at the time of this writing, but may be selected -; with "progzone" -; -; This feature can also easily detect false hangups. The symptoms of this is -; being disconnected in the middle of a call for no reason. -; -;callprogress=yes -;progzone=us -; -; FXO (FXS signalled) devices must have a timeout to determine whe there was a -; hangup before the line was answered. This value can be tweaked to shorten -; how long it takes before Zap considers a non-ringing line to have hungup. -; -;ringtimeout=8000 -; -; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF -; -;pulsedial=yes -; -; For fax detection, uncomment one of the following lines. The default is *OFF* -; -;faxdetect=both -;faxdetect=incoming -;faxdetect=outgoing -;faxdetect=no -; -; Select which class of music to use for music on hold. If not specified -; then the default will be used. -; -;musiconhold=default -; -; PRI channels can have an idle extension and a minunused number. So long as -; at least "minunused" channels are idle, chan_zap will try to call "idledial" -; on them, and then dump them into the PBX in the "idleext" extension (which -; is of the form exten@context). When channels are needed the "idle" calls -; are disconnected (so long as there are at least "minidle" calls still -; running, of course) to make more channels available. The primary use of -; this is to create a dynamic service, where idle channels are bundled through -; multilink PPP, thus more efficiently utilizing combined voice/data services -; than conventional fixed mappings/muxings. -; -;idledial=6999 -;idleext=6999@dialout -;minunused=2 -;minidle=1 -; -; Configure jitter buffers in zapata (each one is 20ms, default is 4) -; -;jitterbuffers=4 -; -; You can define your own custom ring cadences here. You can define up to 8 -; pairs. If the silence is negative, it indicates where the callerid spill is -; to be placed. Also, if you define any custom cadences, the default cadences -; will be turned off. -; -; Syntax is: cadence=ring,silence[,ring,silence[...]] -; -; These are the default cadences: -; -;cadence=125,125,2000,-4000 -;cadence=250,250,500,1000,250,250,500,-4000 -;cadence=125,125,125,125,125,-4000 -;cadence=1000,500,2500,-5000 -; -; Each channel consists of the channel number or range. It inherits the -; parameters that were specified above its declaration. -; -; For GR-303, CRV's are created like channels except they must start with the -; trunk group followed by a colon, e.g.: -; -; crv => 1:1 -; crv => 2:1-2,5-8 -; -; -;callerid="Green Phone"<(256) 428-6121> -;channel => 1 -;callerid="Black Phone"<(256) 428-6122> -;channel => 2 -;callerid="CallerID Phone" <(256) 428-6123> -;callerid="CallerID Phone" <(630) 372-1564> -;callerid="CallerID Phone" <(256) 704-4666> -;channel => 3 -;callerid="Pac Tel Phone" <(256) 428-6124> -;channel => 4 -;callerid="Uniden Dead" <(256) 428-6125> -;channel => 5 -;callerid="Cortelco 2500" <(256) 428-6126> -;channel => 6 -;callerid="Main TA 750" <(256) 428-6127> -;channel => 44 -; -; For example, maybe we have some other channels which start out in a -; different context and use E & M signalling instead. -; -;context=remote -;sigalling=em -;channel => 15 -;channel => 16 - -;signalling=em_w -; -; All those in group 0 I'll use for outgoing calls -; -; Strip most significant digit (9) before sending -; -;stripmsd=1 -;callerid=asreceived -;group=0 -;signalling=fxs_ls -;channel => 45 - -;signalling=fxo_ls -;group=1 -;callerid="Joe Schmoe" <(256) 428-6131> -;channel => 25 -;callerid="Megan May" <(256) 428-6132> -;channel => 26 -;callerid="Suzy Queue" <(256) 428-6233> -;channel => 27 -;callerid="Larry Moe" <(256) 428-6234> -;channel => 28 -; -; Sample PRI (CPE) config: Specify the switchtype, the signalling as either -; pri_cpe or pri_net for CPE or Network termination, and generally you will -; want to create a single "group" for all channels of the PRI. -; -; switchtype = national -; signalling = pri_cpe -; group = 2 -; channel => 1-23 - -; - -; Used for distintive ring support for x100p. -; You can see the dringX patterns is to set any one of the dringXcontext fields -; and they will be printed on the console when an inbound call comes in. -; -;dring1=95,0,0 -;dring1context=internal1 -;dring2=325,95,0 -;dring2context=internal2 -; If no pattern is matched here is where we go. -;context=default -;channel => 1 - |