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Merged revisions 42716 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines

Spelling/grammar fixes (Issue 7929)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Tilghman Lesher 19 years ago
parent ef879455f5
commit 091e1aed8d
20 changed files with 59 additions and 59 deletions
Show Stats Download Patch File Download Diff File

6
configs/alsa.conf.sample
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@ -27,7 +27,7 @@ extension=s
;
;mohinterpret=default
;
; Silence supression can be enabled when sound is over a certain threshold.
; Silence suppression 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
@ -49,11 +49,11 @@ extension=s
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementation are currenlty available - "fixed"
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

2
configs/cdr.conf.sample
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@ -35,7 +35,7 @@
;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
; records. Posting can either occur 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

4
configs/dundi.conf.sample
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@ -89,7 +89,7 @@ autokill=yes
;
; '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
; valid to receive multiple responses 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.
;
@ -144,7 +144,7 @@ autokill=yes
; '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.
; redundancy without sacrificing performance.
;
; include - Includes this peer when searching a particular context
; for lookup (set "all" to perform all lookups with that

2
configs/enum.conf.sample
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@ -11,7 +11,7 @@
;
search => e164.arpa
;
; If you'd like to use the E.164.org public ENUM registery in addition
; If you'd like to use the E.164.org public ENUM registry in addition
; to the official e164.arpa one, uncomment the following line
;
;search => e164.org

4
configs/extensions.ael.sample
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@ -63,7 +63,7 @@ globals {
//
// 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.
// preceded 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
@ -189,7 +189,7 @@ context iaxtel700 {
};
//
// The SWITCH statement permits a server to share the dialplain with
// The SWITCH statement permits a server to share the dialplan 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.

6
configs/extensions.conf.sample
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@ -110,7 +110,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;
; 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.
; preceded 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
@ -228,7 +228,7 @@ include => dundi-e164-lookup
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplain with
; The SWITCH statement permits a server to share the dialplan 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.
@ -432,7 +432,7 @@ 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.
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test

2
configs/features.conf.sample
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@ -22,7 +22,7 @@ context => parkedcalls ; Which context parked calls are in
; as long as the class is not set on the channel directly
; using Set(CHANNEL(musicclass)=whatever) in the dialplan
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
; (default is 3 seconds)
;xfersound = beep ; to indicate an attended transfer is complete
;xferfailsound = beeperr ; to indicate a failed transfer

4
configs/iax.conf.sample
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@ -231,7 +231,7 @@ autokill=yes
;
; 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.
; disabled - Disable the consideration of codec preference altogether.
; (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
@ -359,7 +359,7 @@ host=216.207.245.47
;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
; results to reduce falsely detected LAGGED hosts
; Default: Off
;qualifyfreqok = 60000 ; how frequently to ping the peer when
; everything seems to be ok, in milliseconds

4
configs/mgcp.conf.sample
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@ -21,11 +21,11 @@
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a MGCP
; channel. Two implementations are currenlty available - "fixed"
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

22
configs/misdn.conf.sample
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@ -89,7 +89,7 @@ l1watcher_timeout=0
;
stop_tone_after_first_digit=yes
; wether to append overlapdialed Digits to Extension or not
; whether to append overlapdialed Digits to Extension or not
;
; default value: yes
;
@ -97,7 +97,7 @@ append_digits2exten=yes
;;; CRYPTION STUFF
; Wether to look for dynamic crypting attempt
; Whether to look for dynamic crypting attempt
;
; default value: no
;
@ -119,7 +119,7 @@ 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
; the sections are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name,
; chan_misdn tries every port in this section to find a
; new free channel
@ -149,7 +149,7 @@ language=en
musicclass=default
;
; Either if we should produce DTMF Tones ourselve
; Either if we should produce DTMF Tones ourselves
;
senddtmf=yes
@ -181,7 +181,7 @@ internationalprefix=00
rxgain=0
txgain=0
; some telcos espacially in NL seem to need this set to yes, also in
; some telcos especially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
@ -192,7 +192,7 @@ te_choose_channel=no
;
; This option defines, if chan_misdn should check the L1 on a PMP
; before makeing a group call on it. The L1 may go down for PMP Ports
; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
; as well, since chan_misdn has no chance to distinguish if the L1 is down
@ -298,7 +298,7 @@ nodialtone=no
; Pickup and Callgroup
;
; deafult values: not set = 0
; default values: not set = 0
; range: 0-63
;
;callgroup=1
@ -312,7 +312,7 @@ nodialtone=no
; s=0, p=0 -> callerid presented not screened
; s=1, p=1 -> callerid presented but screened (the remote end does not see it!)
;
; defaule values s=-1, p=-1
; default values s=-1, p=-1
presentation=-1
screen=-1
@ -364,7 +364,7 @@ hdlc=no
;
; defines the maximum amount of incoming calls per port for
; this group. Calls which exceed the maximum will be marked with
; the channel varible MAX_OVERFLOW. It will contain the amount of
; the channel variable MAX_OVERFLOW. It will contain the amount of
; overflowed calls
;
max_incoming=-1
@ -392,11 +392,11 @@ ports=3
[first_extern]
; again port defs
ports=4
; again a context for incomming calls
; again a context for incoming calls
context=Extern1
; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
; here you can give a comma seperated list or simply an '*' for
; here you can give a comma separated list or simply an '*' for
; any msn.
msns=*

2
configs/osp.conf.sample
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@ -10,7 +10,7 @@
;
[general]
;
; Should hardware accelleration be enabled? May not be changed
; Should hardware acceleration be enabled? May not be changed
; on a reload.
;
;accelerate=yes

4
configs/oss.conf.sample
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@ -58,11 +58,11 @@
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
; channel. Two implementations are currenlty available - "fixed"
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

4
configs/phone.conf.sample
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@ -30,7 +30,7 @@ format=slinear
;
echocancel=medium
;
; You can optionally use VAD/CNG silence supression
; You can optionally use VAD/CNG silence suppression
;
;silencesupression=yes
;
@ -40,7 +40,7 @@ echocancel=medium
;
; 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
; add txgain or rxgain line before device line. But remember, 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%).
;

4
configs/queues.conf.sample
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@ -24,7 +24,7 @@ persistentmembers = yes
; probably more along the lines of how a queue should work and
; in most cases, you will want to enable this behavior. If you
; do not specify or comment out this option, it will default to no
; to keep backward compatability with the old behavior.
; to keep backward compatibility with the old behavior.
;
autofill = yes
;
@ -35,7 +35,7 @@ autofill = yes
; the concept of "joining/mixing" the in/out files now goes away
; when this is enabled. You can set the default type for all queues
; here, and then also change monitor-type for individual queues within
; queue by using the same configuation parameter within a queue
; queue by using the same configuration parameter within a queue
; configuration block. If you do not specify or comment out this option,
; it will default to the old 'Monitor' behavior to keep backward
; compatibility.

12
configs/sip.conf.sample
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@ -60,7 +60,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outoing registration
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
@ -294,7 +294,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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 spiteof it having expired
; 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
@ -353,11 +353,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currenlty available - "fixed"
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
@ -465,7 +465,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you propably have NAT problems.
; If you have one-way audio, you probably 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
@ -566,7 +566,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;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
;defaultip=192.168.0.60 ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0

4
configs/skinny.conf.sample
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@ -25,7 +25,7 @@ keepalive=120
;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a
@ -48,7 +48,7 @@ keepalive=120
; Typical config for a 7910
;[duba] ; Device name
;device=SEP0007EB463101 ; Offical identifier
;device=SEP0007EB463101 ; Official identifier
;version=P002F202 ; Firmware version identifier
;host=192.168.1.144
;permit=192.168.0/24 ; Optional, used for authentication

4
configs/smdi.conf.sample
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@ -11,7 +11,7 @@
;twostopbits = no
; Character size or bit length is the size of each character sent accross the
; Character size or bit length is the size of each character sent across the
; link. Character size can be 7 or 8. The default is 7.
;charsize = 7
@ -34,7 +34,7 @@
; Occasionally Asterisk and the SMDI switch may become out of sync. If this
; happens, Asterisk will appear one or several calls behind as it processes
; voicemail requests. To prevent this from hapening adjust the msgexpirytime.
; voicemail requests. To prevent this from happening, adjust the msgexpirytime.
; This will make Asterisk discard old SMDI messages that have not yet been
; processed. The default expiry time is 30000 milliseconds.

10
configs/voicemail.conf.sample
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@ -41,7 +41,7 @@ attach=yes
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many miliseconds to skip forward/back when rew/ff in message playback
; How many milliseconds to skip forward/back when rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
@ -54,7 +54,7 @@ maxlogins=3
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this. It can also be set to 'smdi' to use
; smdi for external notification. If it is 'smdi', smdiport should
; be set to a valid port as specfied in smdi.conf.
; be set to a valid port as specified in smdi.conf.
;externnotify=/usr/bin/myapp
;smdiport=/dev/ttyS0
@ -87,7 +87,7 @@ maxlogins=3
; 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
; just the CIDNAME, if it is not null, otherwise 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
;
@ -147,11 +147,11 @@ emaildateformat=%A, %B %d, %Y at %r
; 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.
; considered to be unchangeable.
;
; 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.
; overridden in the per-mailbox settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages below. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email

2
configs/vpb.conf.sample
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@ -42,7 +42,7 @@
; txhwgain => Transmit hardware gain (-12 => 12)
; rxhwgain => Receive Hardware gain (-12 => 12)
;
; These are advanced settings and only mentioned for fullnes.
; These are advanced settings and only mentioned for completeness.
; bal1 => Hybrid balance codec register 1
; bal2 => Hybrid balance codec register 2
; bal3 => Hybrid balance codec register 3

16
configs/zapata.conf.sample
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@ -207,7 +207,7 @@ signalling=fxo_ls
; format. If you only specify 'signalling', then it will be the format for
; both inbound and outbound.
;
; signaling=featdmf
; signalling=featdmf
; outsignalling=featb
;
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
@ -229,7 +229,7 @@ signalling=fxo_ls
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; How long generated tones (DTMF and MF) will be played on the channel
; (in miliseconds)
; (in milliseconds)
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines
@ -316,7 +316,7 @@ callreturn=yes
; 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.
; if voicemail received in mailbox in the specified voicemail context.
;
; for default voicemail context, the example below is fine:
;
@ -380,7 +380,7 @@ 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
; you can answer it by picking up and dialling *8#. For simple offices, just
; make these both the same. Groups range from 0 to 63.
;
callgroup=1
@ -419,7 +419,7 @@ immediate=no
;
; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
; basis if you would like that channel to behave like an SMDI message desk.
; The SMDI port specfied should have already been defined in smdi.conf. The
; The SMDI port specified should have already been defined in smdi.conf. The
; default port is /dev/ttyS0.
;
;usesmdi=yes
@ -547,11 +547,11 @@ immediate=no
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a ZAP
; channel. Two implementations are currenlty available - "fixed"
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
@ -641,7 +641,7 @@ immediate=no
;
; Used for distintive ring support for x100p.
; Used for distinctive 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.
;

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