diff options
| -rw-r--r-- | configs/alsa.conf.sample | 6 | ||||
| -rw-r--r-- | configs/cdr.conf.sample | 2 | ||||
| -rw-r--r-- | configs/dundi.conf.sample | 4 | ||||
| -rw-r--r-- | configs/enum.conf.sample | 2 | ||||
| -rw-r--r-- | configs/extensions.ael.sample | 4 | ||||
| -rw-r--r-- | configs/extensions.conf.sample | 6 | ||||
| -rw-r--r-- | configs/features.conf.sample | 2 | ||||
| -rw-r--r-- | configs/iax.conf.sample | 4 | ||||
| -rw-r--r-- | configs/mgcp.conf.sample | 4 | ||||
| -rw-r--r-- | configs/misdn.conf.sample | 22 | ||||
| -rw-r--r-- | configs/osp.conf.sample | 2 | ||||
| -rw-r--r-- | configs/oss.conf.sample | 4 | ||||
| -rw-r--r-- | configs/phone.conf.sample | 4 | ||||
| -rw-r--r-- | configs/queues.conf.sample | 4 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 12 | ||||
| -rw-r--r-- | configs/skinny.conf.sample | 4 | ||||
| -rw-r--r-- | configs/smdi.conf.sample | 4 | ||||
| -rw-r--r-- | configs/voicemail.conf.sample | 10 | ||||
| -rw-r--r-- | configs/vpb.conf.sample | 2 | ||||
| -rw-r--r-- | configs/zapata.conf.sample | 16 |
20 files changed, 59 insertions, 59 deletions
diff --git a/configs/alsa.conf.sample b/configs/alsa.conf.sample index 9b036a9d9..f55030618 100644 --- a/configs/alsa.conf.sample +++ b/configs/alsa.conf.sample @@ -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. diff --git a/configs/cdr.conf.sample b/configs/cdr.conf.sample index eb3ababcd..4e4c3d43c 100644 --- a/configs/cdr.conf.sample +++ b/configs/cdr.conf.sample @@ -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 diff --git a/configs/dundi.conf.sample b/configs/dundi.conf.sample index a3c8c77d9..a1e999726 100644 --- a/configs/dundi.conf.sample +++ b/configs/dundi.conf.sample @@ -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 diff --git a/configs/enum.conf.sample b/configs/enum.conf.sample index 8d7054a24..b7e02bfe5 100644 --- a/configs/enum.conf.sample +++ b/configs/enum.conf.sample @@ -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 diff --git a/configs/extensions.ael.sample b/configs/extensions.ael.sample index 6c5706bbc..67aac81f0 100644 --- a/configs/extensions.ael.sample +++ b/configs/extensions.ael.sample @@ -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. diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample index e88404c5e..864644c2f 100644 --- a/configs/extensions.conf.sample +++ b/configs/extensions.conf.sample @@ -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 diff --git a/configs/features.conf.sample b/configs/features.conf.sample index a95a1238c..0cdfd0e67 100644 --- a/configs/features.conf.sample +++ b/configs/features.conf.sample @@ -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 diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample index f616e4b21..234d18e9e 100644 --- a/configs/iax.conf.sample +++ b/configs/iax.conf.sample @@ -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 diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample index c80104e32..288065355 100644 --- a/configs/mgcp.conf.sample +++ b/configs/mgcp.conf.sample @@ -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. diff --git a/configs/misdn.conf.sample b/configs/misdn.conf.sample index 21335b0c0..e077e9089 100644 --- a/configs/misdn.conf.sample +++ b/configs/misdn.conf.sample @@ -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=* diff --git a/configs/osp.conf.sample b/configs/osp.conf.sample index 75545716f..edd8af1e3 100644 --- a/configs/osp.conf.sample +++ b/configs/osp.conf.sample @@ -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 diff --git a/configs/oss.conf.sample b/configs/oss.conf.sample index ec42da771..fb2d0281a 100644 --- a/configs/oss.conf.sample +++ b/configs/oss.conf.sample @@ -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. diff --git a/configs/phone.conf.sample b/configs/phone.conf.sample index 82909f2e6..649edc7bb 100644 --- a/configs/phone.conf.sample +++ b/configs/phone.conf.sample @@ -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%). ; diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample index 40098606f..537277235 100644 --- a/configs/queues.conf.sample +++ b/configs/queues.conf.sample @@ -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. diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index b04c60e77..f931780b9 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -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 diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample index c131c2963..54e304785 100644 --- a/configs/skinny.conf.sample +++ b/configs/skinny.conf.sample @@ -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 diff --git a/configs/smdi.conf.sample b/configs/smdi.conf.sample index aa58bfbdd..0325eba48 100644 --- a/configs/smdi.conf.sample +++ b/configs/smdi.conf.sample @@ -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. diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample index 3d59f4f2f..53d16278b 100644 --- a/configs/voicemail.conf.sample +++ b/configs/voicemail.conf.sample @@ -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 diff --git a/configs/vpb.conf.sample b/configs/vpb.conf.sample index 480deee06..4a9b0b36a 100644 --- a/configs/vpb.conf.sample +++ b/configs/vpb.conf.sample @@ -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 diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample index 359df8bd8..ea13db23e 100644 --- a/configs/zapata.conf.sample +++ b/configs/zapata.conf.sample @@ -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. ; |