Merged revisions 42716 via svnmerge from
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
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@ -27,7 +27,7 @@ extension=s
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;
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;mohinterpret=default
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;
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; Silence supression can be enabled when sound is over a certain threshold.
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; Silence suppression can be enabled when sound is over a certain threshold.
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; The value for the threshold should probably be between 500 and 2000 or so,
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; but your mileage may vary. Use the echo test to evaluate the best setting.
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;silencesuppression = yes
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@ -49,11 +49,11 @@ extension=s
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usualy sent from exotic devices
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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; channel. Two implementation are currenlty available - "fixed"
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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@ -35,7 +35,7 @@
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;time=300
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; The CDR engine uses the internal asterisk scheduler to determine when to post
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; records. Posting can either occure inside the scheduler thread, or a new
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; records. Posting can either occur inside the scheduler thread, or a new
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; thread can be spawned for the submission of every batch. For small batches,
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; it might be acceptable to just use the scheduler thread, so set this to "yes".
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; For large batches, say anything over size=10, a new thread is recommended, so
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@ -89,7 +89,7 @@ autokill=yes
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;
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; 'weight' is the weight to use for the responses provided from this
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; mapping. The number must be >= 0 and < 60000. Since it is totally
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; valid to receive multiple reponses to a query, responses received
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; valid to receive multiple responses to a query, responses received
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; with a lower weight are tried first. Note that the weight has a
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; special meaning in the e164 context - see the GPA for more details.
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;
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@ -144,7 +144,7 @@ autokill=yes
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; 'tertiary' or 'quartiary'. In large systems, it is beneficial
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; to only query one up-stream host in order to maximize caching
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; value. Adding one with primary and one with secondary gives you
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; redundancy without sacraficing performance.
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; redundancy without sacrificing performance.
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;
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; include - Includes this peer when searching a particular context
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; for lookup (set "all" to perform all lookups with that
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@ -11,7 +11,7 @@
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;
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search => e164.arpa
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;
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; If you'd like to use the E.164.org public ENUM registery in addition
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; If you'd like to use the E.164.org public ENUM registry in addition
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; to the official e164.arpa one, uncomment the following line
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;
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;search => e164.org
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@ -63,7 +63,7 @@ globals {
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//
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// For example the extension _NXXXXXX would match normal 7 digit dialings,
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// while _1NXXNXXXXXX would represent an area code plus phone number
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// preceeded by a one.
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// preceded by a one.
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//
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// Each step of an extension is ordered by priority, which must
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// always start with 1 to be considered a valid extension. The priority
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@ -189,7 +189,7 @@ context iaxtel700 {
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};
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//
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// The SWITCH statement permits a server to share the dialplain with
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// The SWITCH statement permits a server to share the dialplan with
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// another server. Use with care: Reciprocal switch statements are not
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// allowed (e.g. both A -> B and B -> A), and the switched server needs
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// to be on-line or else dialing can be severly delayed.
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@ -110,7 +110,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
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;
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; For example the extension _NXXXXXX would match normal 7 digit dialings,
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; while _1NXXNXXXXXX would represent an area code plus phone number
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; preceeded by a one.
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; preceded by a one.
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;
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; Each step of an extension is ordered by priority, which must
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; always start with 1 to be considered a valid extension. The priority
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@ -228,7 +228,7 @@ include => dundi-e164-lookup
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exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
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;
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; The SWITCH statement permits a server to share the dialplain with
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; The SWITCH statement permits a server to share the dialplan with
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; another server. Use with care: Reciprocal switch statements are not
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; allowed (e.g. both A -> B and B -> A), and the switched server needs
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; to be on-line or else dialing can be severly delayed.
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@ -432,7 +432,7 @@ exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
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exten => 500,n,Goto(s,6) ; Return to the start over message.
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;
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; Create an extension, 600, for evaulating echo latency.
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; Create an extension, 600, for evaluating echo latency.
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;
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exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
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exten => 600,n,Echo ; Do the echo test
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@ -22,7 +22,7 @@ context => parkedcalls ; Which context parked calls are in
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; as long as the class is not set on the channel directly
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; using Set(CHANNEL(musicclass)=whatever) in the dialplan
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;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call
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;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
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; (default is 3 seconds)
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;xfersound = beep ; to indicate an attended transfer is complete
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;xferfailsound = beeperr ; to indicate a failed transfer
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@ -231,7 +231,7 @@ autokill=yes
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;
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; caller - Consider the callers preferred order ahead of the host's.
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; host - Consider the host's preferred order ahead of the caller's.
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; disabled - Disable the consideration of codec preference alltogether.
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; disabled - Disable the consideration of codec preference altogether.
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; (this is the original behaviour before preferences were added)
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; reqonly - Same as disabled, only do not consider capabilities if
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; the requested format is not available the call will only
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@ -359,7 +359,7 @@ host=216.207.245.47
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;mask=255.255.255.255
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;qualify=yes ; Make sure this peer is alive
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;qualifysmoothing = yes ; use an average of the last two PONG
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; results to reduce falsly detected LAGGED hosts
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; results to reduce falsely detected LAGGED hosts
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; Default: Off
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;qualifyfreqok = 60000 ; how frequently to ping the peer when
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; everything seems to be ok, in milliseconds
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@ -21,11 +21,11 @@
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usualy sent from exotic devices
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a MGCP
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; channel. Two implementations are currenlty available - "fixed"
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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@ -89,7 +89,7 @@ l1watcher_timeout=0
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;
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stop_tone_after_first_digit=yes
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; wether to append overlapdialed Digits to Extension or not
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; whether to append overlapdialed Digits to Extension or not
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;
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; default value: yes
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;
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@ -97,7 +97,7 @@ append_digits2exten=yes
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;;; CRYPTION STUFF
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; Wether to look for dynamic crypting attempt
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; Whether to look for dynamic crypting attempt
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;
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; default value: no
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;
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@ -119,7 +119,7 @@ crypt_keys=test,muh
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; users sections:
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;
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; name your sections as you which but not "general" !
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; the secions are Groups, you can dial out in extensions.conf
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; the sections are Groups, you can dial out in extensions.conf
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; with Dial(mISDN/g:extern/101) where extern is a section name,
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; chan_misdn tries every port in this section to find a
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; new free channel
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@ -149,7 +149,7 @@ language=en
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musicclass=default
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;
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; Either if we should produce DTMF Tones ourselve
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; Either if we should produce DTMF Tones ourselves
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;
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senddtmf=yes
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@ -181,7 +181,7 @@ internationalprefix=00
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rxgain=0
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txgain=0
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; some telcos espacially in NL seem to need this set to yes, also in
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; some telcos especially in NL seem to need this set to yes, also in
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; switzerland this seems to be important
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;
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; default value: no
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@ -192,7 +192,7 @@ te_choose_channel=no
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;
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; This option defines, if chan_misdn should check the L1 on a PMP
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; before makeing a group call on it. The L1 may go down for PMP Ports
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; before making a group call on it. The L1 may go down for PMP Ports
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; so we might need this.
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; But be aware! a broken or plugged off cable might be used for a group call
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; as well, since chan_misdn has no chance to distinguish if the L1 is down
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@ -298,7 +298,7 @@ nodialtone=no
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; Pickup and Callgroup
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;
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; deafult values: not set = 0
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; default values: not set = 0
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; range: 0-63
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;
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;callgroup=1
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@ -312,7 +312,7 @@ nodialtone=no
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; s=0, p=0 -> callerid presented not screened
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; s=1, p=1 -> callerid presented but screened (the remote end does not see it!)
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;
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; defaule values s=-1, p=-1
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; default values s=-1, p=-1
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presentation=-1
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screen=-1
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@ -364,7 +364,7 @@ hdlc=no
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;
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; defines the maximum amount of incoming calls per port for
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; this group. Calls which exceed the maximum will be marked with
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; the channel varible MAX_OVERFLOW. It will contain the amount of
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; the channel variable MAX_OVERFLOW. It will contain the amount of
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; overflowed calls
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;
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max_incoming=-1
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@ -392,11 +392,11 @@ ports=3
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[first_extern]
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; again port defs
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ports=4
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; again a context for incomming calls
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; again a context for incoming calls
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context=Extern1
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; msns for te ports, listen on those numbers on the above ports, and
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; indicate the incoming calls to asterisk
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; here you can give a comma seperated list or simply an '*' for
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; here you can give a comma separated list or simply an '*' for
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; any msn.
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msns=*
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@ -10,7 +10,7 @@
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;
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[general]
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;
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; Should hardware accelleration be enabled? May not be changed
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; Should hardware acceleration be enabled? May not be changed
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; on a reload.
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;
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;accelerate=yes
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@ -58,11 +58,11 @@
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usualy sent from exotic devices
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
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; channel. Two implementations are currenlty available - "fixed"
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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@ -30,7 +30,7 @@ format=slinear
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;
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echocancel=medium
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;
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; You can optionally use VAD/CNG silence supression
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; You can optionally use VAD/CNG silence suppression
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;
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;silencesupression=yes
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;
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@ -40,7 +40,7 @@ echocancel=medium
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;
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; You can set txgain and rxgain for each device in the same way as context.
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; If you want to change default gain value (1.0 =~ 100%) for device, simple
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; add txgain or rxgain line before device line. But rememeber, if you change
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; add txgain or rxgain line before device line. But remember, if you change
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; volume all cards listed below will be affected by these values. You can
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; use float values (1.0, 0.5, 2.0) or percentage values (100%, 150%, 50%).
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;
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@ -24,7 +24,7 @@ persistentmembers = yes
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; probably more along the lines of how a queue should work and
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; in most cases, you will want to enable this behavior. If you
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; do not specify or comment out this option, it will default to no
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; to keep backward compatability with the old behavior.
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; to keep backward compatibility with the old behavior.
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;
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autofill = yes
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;
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@ -35,7 +35,7 @@ autofill = yes
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; the concept of "joining/mixing" the in/out files now goes away
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; when this is enabled. You can set the default type for all queues
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; here, and then also change monitor-type for individual queues within
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; queue by using the same configuation parameter within a queue
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; queue by using the same configuration parameter within a queue
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; configuration block. If you do not specify or comment out this option,
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; it will default to the old 'Monitor' behavior to keep backward
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; compatibility.
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@ -60,7 +60,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;maxexpiry=3600 ; Maximum allowed time of incoming registrations
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; and subscriptions (seconds)
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outoing registration
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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; Defaults to 100 ms
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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@ -294,7 +294,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; For non-realtime peers, when their registration expires, the
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; information will _not_ be removed from memory or the Asterisk database
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; if you attempt to place a call to the peer, the existing information
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; will be used in spiteof it having expired
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; will be used in spite of it having expired
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;
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; For realtime peers, when the peer is retrieved from realtime storage,
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; the registration information will be used regardless of whether
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@ -353,11 +353,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
|
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; big jumps in/broken timestamps, usualy sent from exotic devices
|
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; big jumps in/broken timestamps, usually sent from exotic devices
|
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; and programs. Defaults to 1000.
|
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|
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
||||
; channel. Two implementations are currenlty available - "fixed"
|
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; 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.
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@ -465,7 +465,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;
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; For local phones, type=friend works most of the time
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;
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; If you have one-way audio, you propably have NAT problems.
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; If you have one-way audio, you probably have NAT problems.
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; If Asterisk is on a public IP, and the phone is inside of a NAT device
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; you will need to configure nat option for those phones.
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; Also, turn on qualify=yes to keep the nat session open
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@ -566,7 +566,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;
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;callgroup=1,3-4 ; We are in caller groups 1,3,4
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;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
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;defaultip=192.168.0.60 ; IP address to use if peer has not registred
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;defaultip=192.168.0.60 ; IP address to use if peer has not registered
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;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
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;permit=192.168.0.60/255.255.255.0
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||||
|
|
|
@ -25,7 +25,7 @@ keepalive=120
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|||
|
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;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.
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||||
|
||||
;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a
|
||||
|
@ -48,7 +48,7 @@ keepalive=120
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|||
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||||
; Typical config for a 7910
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||||
;[duba] ; Device name
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||||
;device=SEP0007EB463101 ; Offical identifier
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||||
;device=SEP0007EB463101 ; Official identifier
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||||
;version=P002F202 ; Firmware version identifier
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;host=192.168.1.144
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;permit=192.168.0/24 ; Optional, used for authentication
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|
|
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@ -11,7 +11,7 @@
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;twostopbits = no
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; Character size or bit length is the size of each character sent accross the
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; Character size or bit length is the size of each character sent across the
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||||
; link. Character size can be 7 or 8. The default is 7.
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;charsize = 7
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||||
|
@ -34,7 +34,7 @@
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|||
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||||
; Occasionally Asterisk and the SMDI switch may become out of sync. If this
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||||
; happens, Asterisk will appear one or several calls behind as it processes
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||||
; voicemail requests. To prevent this from hapening adjust the msgexpirytime.
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||||
; voicemail requests. To prevent this from happening, adjust the msgexpirytime.
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||||
; This will make Asterisk discard old SMDI messages that have not yet been
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||||
; processed. The default expiry time is 30000 milliseconds.
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||||
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||||
|
|
|
@ -41,7 +41,7 @@ attach=yes
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;minmessage=3
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||||
; Maximum length of greetings in seconds
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||||
;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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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.
|
||||
;
|
||||
|
|
Loading…
Reference in New Issue