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make sample config files easier to ready (issue #5371)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
@@ -22,8 +22,9 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
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;eventcmd = yourprogram -yourargs ...
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;
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; Specify a spool directory for the event files. This setting is required if you want the app to be useful.
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; Event files written to the spool directory will be of the template event-XXXXXX, where XXXXXX is a random
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; Specify a spool directory for the event files. This setting is required
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; if you want the app to be useful. Event files written to the spool
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; directory will be of the template event-XXXXXX, where XXXXXX is a random
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; and unique alphanumeric string.
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;
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; Default is none, and the events will be dropped on the floor.
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@@ -32,8 +33,9 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
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eventspooldir = /tmp
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;
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; The alarmreceiver app can either log the events one-at-a-time to individual files in the spool
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; directory, or it can store them until the caller disconnects and write them all to one file.
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; The alarmreceiver app can either log the events one-at-a-time to individual
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; files in the spool directory, or it can store them until the caller
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; disconnects and write them all to one file.
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;
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; The default setting for logindividualevents is no.
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;
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@@ -41,32 +43,34 @@ eventspooldir = /tmp
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logindividualevents = no
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;
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; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. to 10000 msec. The
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; default is 2000 msec. Note: if you wish to test the receiver by entering digits manually, set this
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; to a reasonable time out like 10000 milliseconds.
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; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
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; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
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; receiver by entering digits manually, set this to a reasonable time out
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; like 10000 milliseconds.
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fdtimeout = 2000
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;
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; The timeout for receiving subsequent DTMF digits is adjustable from 110 msec. to 4000 msec. The
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; default is 200 msec. Note: if you wish to test the receiver by entering digits manually, set this
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; to a reasonable time out like 4000 milliseconds.
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; The timeout for receiving subsequent DTMF digits is adjustable from
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; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
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; the receiver by entering digits manually, set this to a reasonable time out
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; like 4000 milliseconds.
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;
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sdtimeout = 200
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;
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; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. The default is 8192
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; This shouldn't need to be messed with, but is included just in case there are problems with
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; signal levels.
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; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
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; The default is 8192. This shouldn't need to be messed with, but is included
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; just in case there are problems with signal levels.
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;
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loudness = 8192
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;
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; The db-family setting allows the user to capture statistics on the number of calls, and the errors
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; the alarm receiver sees. The default is for no db-family name to be defined and the database logging
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; to be turned off.
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; The db-family setting allows the user to capture statistics on the number of
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; calls, and the errors the alarm receiver sees. The default is for no
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; db-family name to be defined and the database logging to be turned off.
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;
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;db-family = yourfamily:
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@@ -12,7 +12,8 @@ complexity => 2
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enhancement => true
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; voice activity detection [true / false]
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; reduces bitrate when no voice detected, used only for CBR (implicit in VBR/ABR)
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; reduces bitrate when no voice detected, used only for CBR
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; (implicit in VBR/ABR)
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vad => true
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; variable bit rate [true / false]
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@@ -52,13 +52,15 @@ clearglobalvars=no
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;
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priorityjumping=no
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;
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; You can include other config files, use the #include command (without the ';')
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; Note that this is different from the "include" command that includes contexts within
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; other contexts. The #include command works in all asterisk configuration files.
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; You can include other config files, use the #include command
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; (without the ';'). Note that this is different from the "include" command
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; that includes contexts within other contexts. The #include command works
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; in all asterisk configuration files.
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;#include "filename.conf"
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; The "Globals" category contains global variables that can be referenced
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; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
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; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
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; variables,
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; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
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;
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[globals]
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@@ -73,10 +75,14 @@ TRUNK=Zap/g2 ; Trunk interface
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; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
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; the specified group. The four possible options are:
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;
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; g: select the lowest-numbered non-busy Zap channel (aka. ascending sequential hunt group).
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; G: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group).
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; r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group).
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; R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group).
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; g: select the lowest-numbered non-busy Zap channel
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; (aka. ascending sequential hunt group).
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; G: select the highest-numbered non-busy Zap channel
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; (aka. descending sequential hunt group).
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; r: use a round-robin search, starting at the next highest channel than last
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; time (aka. ascending rotary hunt group).
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; R: use a round-robin search, starting at the next lowest channel than last
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; time (aka. descending rotary hunt group).
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;
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TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
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;TRUNK=IAX2/user:pass@provider
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@@ -443,11 +449,11 @@ include => demo
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;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
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;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
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; Real extensions would go here. Generally you want real extensions to be 4 or 5
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; digits long (although there is no such requirement) and start with a single
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; digit that is fairly large (like 6 or 7) so that you have plenty of room to
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; overlap extensions and menu options without conflict. You can alias them with
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; names, too and use global variables
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; Real extensions would go here. Generally you want real extensions to be
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; 4 or 5 digits long (although there is no such requirement) and start with a
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; single digit that is fairly large (like 6 or 7) so that you have plenty of
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; room to overlap extensions and menu options without conflict. You can alias
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; them with names, too, and use global variables
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;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
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;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
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@@ -74,11 +74,11 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
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; The jitter buffer's function is to compensate for varying
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; network delay.
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;
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; There are presently two jitterbuffer implementations available for * and chan_iax2;
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; the classic and the new, channel/application independent implementation. These
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; are controlled at compile-time. The new jitterbuffer additionally has support for PLC
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; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost
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; packets.
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; There are presently two jitterbuffer implementations available for Asterisk
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; and chan_iax2; the classic and the new, channel/application independent
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; implementation. These are controlled at compile-time. The new jitterbuffer
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; additionally has support for PLC which greatly improves quality as the
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; jitterbuffer adapts size, and in compensating for lost packets.
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;
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; All the jitter buffer settings except dropcount are in milliseconds.
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; The jitter buffer works for INCOMING audio - the outbound audio
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@@ -90,7 +90,8 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
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; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
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; we don't want to do jitterbuffering on the switch, since the endpoints
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; can each handle this. However, some endpoints may have poor jitterbuffers
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; themselves, so this option will force * to always jitterbuffer, even in this case.
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; themselves, so this option will force * to always jitterbuffer, even in this
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; case.
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; [This option presently applies only to the new jitterbuffer implementation]
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;
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; dropcount: the jitter buffer is sized such that no more than "dropcount"
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@@ -105,15 +106,17 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
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;
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; resyncthreshold: when the jitterbuffer notices a significant change in delay
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; that continues over a few frames, it will resync, assuming that the change in
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; delay was caused by a timestamping mix-up. The threshold for noticing a change
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; in delay is measured as twice the measured jitter plus this resync threshold.
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; Resycning can be disabled by setting this parameter to -1.
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; delay was caused by a timestamping mix-up. The threshold for noticing a
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; change in delay is measured as twice the measured jitter plus this resync
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; threshold.
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; Resyncing can be disabled by setting this parameter to -1.
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; [This option presently applies only to the new jitterbuffer implementation]
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;
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; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer should
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; return in a row. Since some clients do not send CNG/DTX frames to indicate
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; silence, the jitterbuffer will assume silence has begun after returning this
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; many interpolations. This prevents interpolating throughout a long silence.
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; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
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; should return in a row. Since some clients do not send CNG/DTX frames to
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; indicate silence, the jitterbuffer will assume silence has begun after
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; returning this many interpolations. This prevents interpolating throughout
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; a long silence.
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; [This option presently applies only to the new jitterbuffer implementation]
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;
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; maxexcessbuffer: If conditions improve after a period of high jitter,
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@@ -147,11 +150,11 @@ forcejitterbuffer=no
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;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
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; Should we send timestamps for the individual sub-frames within trunk frames?
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; There is a small bandwidth use for these (less than 1kbps/call), but they ensure
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; that frame timestamps get sent end-to-end properly. If both ends of all your trunks
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; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you
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; can probably suppress these. The receiver must also support this feature, although
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; they do not also need to have it enabled.
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; There is a small bandwidth use for these (less than 1kbps/call), but they
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; ensure that frame timestamps get sent end-to-end properly. If both ends of
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; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
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; length for your codecs, you can probably suppress these. The receiver must
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; also support this feature, although they do not also need to have it enabled.
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;
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; trunktimestamps=yes
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;
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@@ -217,22 +220,21 @@ tos=lowdelay
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;
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;mailboxdetail=yes
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;
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; If regcontext is specified, Asterisk will dynamically
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; create and destroy a NoOp priority 1 extension for a given
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; peer who registers or unregisters with us. The actual extension
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; is the 'regexten' parameter of the registering peer or its
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; name if 'regexten' is not provided. More than one regexten may be supplied
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; if they are separated by '&'. Patterns may be used in regexten.
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; If regcontext is specified, Asterisk will dynamically create and destroy
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; a NoOp priority 1 extension for a given peer who registers or unregisters
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; with us. The actual extension is the 'regexten' parameter of the registering
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; peer or its name if 'regexten' is not provided. More than one regexten
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; may be supplied if they are separated by '&'. Patterns may be used in
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; regexten.
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;
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;regcontext=iaxregistrations
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;
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; If we don't get ACK to our NEW within 2000ms, and autokill is set
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; to yes, then we cancel the whole thing (that's enough time for one
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; retransmission only). This is used to keep things from stalling for a long
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; time for a host that is not available, but would be ill advised for bad
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; connections. In addition to 'yes' or 'no' you can also specify a number
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; of milliseconds. See 'qualify' for individual peers to turn on for just
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; a specific peer.
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; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
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; then we cancel the whole thing (that's enough time for one retransmission
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||||
; only). This is used to keep things from stalling for a long time for a host
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; that is not available, but would be ill advised for bad connections. In
|
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; addition to 'yes' or 'no' you can also specify a number of milliseconds.
|
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; See 'qualify' for individual peers to turn on for just a specific peer.
|
||||
;
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||||
autokill=yes
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||||
;
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||||
@@ -274,8 +276,8 @@ autokill=yes
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; has expired based on its registration interval, used the stored
|
||||
; address information regardless. (yes|no)
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||||
; Guest sections for unauthenticated connection attempts. Just
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; specify an empty secret, or provide no secret section.
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; Guest sections for unauthenticated connection attempts. Just specify an
|
||||
; empty secret, or provide no secret section.
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;
|
||||
[guest]
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||||
type=user
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||||
@@ -310,14 +312,13 @@ inkeys=freeworlddialup
|
||||
;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
|
||||
; 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
|
||||
@@ -372,11 +373,10 @@ host=216.207.245.47
|
||||
;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.
|
||||
; 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]
|
||||
@@ -410,3 +410,4 @@ host=216.207.245.47
|
||||
;secret=moofoo
|
||||
;context=default
|
||||
;permit=0.0.0.0/0.0.0.0
|
||||
|
||||
|
||||
@@ -1,25 +1,22 @@
|
||||
;
|
||||
; IAX2 Provisioning Information
|
||||
;
|
||||
; Contains provisioning information for templates
|
||||
; and for specific service entries.
|
||||
; 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 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.
|
||||
; 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
|
||||
; The port number the device should use to bind to. The default is 4569.
|
||||
;
|
||||
;port=4569
|
||||
;
|
||||
@@ -27,14 +24,13 @@
|
||||
;
|
||||
;server=192.168.69.3
|
||||
;
|
||||
; altserver is the BACKUP server for registration and placing calls
|
||||
; in the event the primary server is unavailable.
|
||||
; 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.
|
||||
; 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
|
||||
@@ -78,9 +74,10 @@ tos=lowdelay
|
||||
;
|
||||
;[*]
|
||||
;
|
||||
; If specified, the '*' provisioning is used for all devices which do
|
||||
; not have another provisioning entry within the file. If unspecified, no
|
||||
; 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
|
||||
|
||||
|
||||
@@ -16,7 +16,7 @@ country=us ; default location
|
||||
|
||||
; [example]
|
||||
; description = string
|
||||
; The full name of your country, in English
|
||||
; 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.
|
||||
@@ -31,14 +31,16 @@ country=us ; default location
|
||||
; 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
|
||||
; 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
|
||||
; 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")
|
||||
; Set of tones played with special information messages (e.g., "number is
|
||||
; out of service")
|
||||
; 'name' = tonelist
|
||||
; Every other variable will be available as a shortcut for the "PlayList" command
|
||||
; but will not automaticly be used by Asterisk.
|
||||
; 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.
|
||||
@@ -587,8 +589,8 @@ stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/1
|
||||
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.
|
||||
; Note, though, that South Africa uses two switch types in their network --
|
||||
; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
|
||||
; The former use 383+417 in dial, ringback etc. The latter use 400*33
|
||||
; I've provided both, uncomment the ones you prefer
|
||||
ringcadance = 400,200,400,2000
|
||||
|
||||
@@ -16,10 +16,12 @@
|
||||
; 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).
|
||||
; 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).
|
||||
; This determines whether or not we log generic events to a file
|
||||
; (defaults to yes).
|
||||
;event_log = no
|
||||
;
|
||||
;
|
||||
@@ -44,17 +46,16 @@
|
||||
;
|
||||
; 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.
|
||||
; 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
|
||||
|
||||
@@ -1,23 +1,19 @@
|
||||
;
|
||||
; AMI - The Asterisk Manager Interface
|
||||
;
|
||||
; Third party application call management support
|
||||
; and PBX event supervision
|
||||
; Third party application call management support and PBX event supervision
|
||||
;
|
||||
; This configuration file is read every time someone
|
||||
; logs in
|
||||
; This configuration file is read every time someone logs in
|
||||
;
|
||||
; Use the "show manager commands" at the CLI to list
|
||||
; availabale manager commands and their authorization
|
||||
; levels.
|
||||
; 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
|
||||
; ---------------------------- 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
|
||||
|
||||
@@ -1,6 +1,5 @@
|
||||
;
|
||||
; Configuration file for MeetMe simple conference rooms
|
||||
; for Asterisk of course.
|
||||
; Configuration file for MeetMe simple conference rooms for Asterisk of course.
|
||||
;
|
||||
; This configuration file is read every time you call app meetme()
|
||||
;
|
||||
@@ -10,3 +9,4 @@
|
||||
;
|
||||
;conf => 1234
|
||||
;conf => 2345,9938
|
||||
|
||||
|
||||
@@ -45,7 +45,8 @@
|
||||
;
|
||||
;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.
|
||||
;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
|
||||
|
||||
|
||||
@@ -7,11 +7,12 @@
|
||||
[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.
|
||||
; 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
|
||||
|
||||
@@ -26,7 +26,8 @@ directory=/var/lib/asterisk/mohmp3
|
||||
;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
|
||||
; mpg123 on Solaris does not always exit properly; madplay may be a better
|
||||
; choice
|
||||
;[solaris]
|
||||
;mode=custom
|
||||
;directory=/var/lib/asterisk/mohmp3
|
||||
|
||||
@@ -9,8 +9,8 @@
|
||||
;
|
||||
persistentmembers = yes
|
||||
;
|
||||
; Note that a timeout to fail out of a queue may be passed as part of application call
|
||||
; from extensions.conf:
|
||||
; 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)
|
||||
|
||||
@@ -43,7 +43,8 @@ persistentmembers = yes
|
||||
;strategy = ringall
|
||||
;
|
||||
; Second settings for service level (default 0)
|
||||
; Used for service level statistics (calls answered within service level time frame)
|
||||
; 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
|
||||
@@ -94,7 +95,8 @@ persistentmembers = yes
|
||||
|
||||
;
|
||||
; 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
|
||||
; If this is non-zero, then we announce the seconds as well as the minutes
|
||||
; rounded to this value.
|
||||
;
|
||||
; announce-round-seconds = 10
|
||||
;
|
||||
@@ -119,26 +121,29 @@ persistentmembers = yes
|
||||
; To enable monitoring, simply specify "monitor-format"; it will be disabled
|
||||
; otherwise.
|
||||
;
|
||||
; You can specify the monitor filename with by calling Set(MONITOR_FILENAME=foo)
|
||||
; Otherwise it will use ${UNIQUEID}
|
||||
; You can specify the monitor filename with by calling
|
||||
; Set(MONITOR_FILENAME=foo)
|
||||
; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
|
||||
;
|
||||
; monitor-format = gsm|wav|wav49
|
||||
;
|
||||
; If you wish to have the two files joined together when the call ends set this to yes
|
||||
; 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:
|
||||
; 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
|
||||
; 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'
|
||||
; 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
|
||||
;
|
||||
@@ -155,14 +160,15 @@ persistentmembers = yes
|
||||
;
|
||||
; 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.
|
||||
; 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.
|
||||
; 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
|
||||
;
|
||||
|
||||
@@ -108,12 +108,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
;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.
|
||||
; 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
|
||||
;
|
||||
@@ -121,12 +120,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
; 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)
|
||||
; 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.
|
||||
; host is either a host name defined in DNS or the name of a section defined
|
||||
; below.
|
||||
;
|
||||
; Examples:
|
||||
;
|
||||
@@ -137,12 +136,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
;
|
||||
;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 define
|
||||
; 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
|
||||
; 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
|
||||
@@ -151,9 +151,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
; Default is 10 tries
|
||||
;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.
|
||||
;----------------------------------------- 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
|
||||
@@ -176,10 +176,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
;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 firewalls support of SIP+RTP ports.
|
||||
; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf
|
||||
; 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
|
||||
@@ -242,7 +242,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
; 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
|
||||
;
|
||||
@@ -341,6 +341,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
;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]
|
||||
|
||||
@@ -10,8 +10,8 @@ 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.
|
||||
; 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
|
||||
@@ -28,13 +28,12 @@ maxsilence=10
|
||||
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:
|
||||
; 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:
|
||||
; 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
|
||||
@@ -54,13 +53,15 @@ maxlogins=3
|
||||
;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
|
||||
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
|
||||
; VM_CIDNAME, VM_DATE
|
||||
;
|
||||
; Note: The emailbody config row can be up to 512 characters due to a limitation in
|
||||
; asterisk config files.
|
||||
; 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, else just the CIDNUM, or "an unknown caller" if they are both null.
|
||||
; 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.
|
||||
@@ -69,7 +70,8 @@ maxlogins=3
|
||||
;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
|
||||
; 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
|
||||
@@ -93,7 +95,8 @@ emaildateformat=%A, %B %d, %Y at %r
|
||||
; variable substitution is done on the values below.
|
||||
;
|
||||
; Supported values:
|
||||
; 'filename' filename of a soundfile (single ticks around the filename required)
|
||||
; '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, ...)
|
||||
@@ -105,8 +108,10 @@ emaildateformat=%A, %B %d, %Y at %r
|
||||
; 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)
|
||||
; 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
|
||||
;
|
||||
;
|
||||
@@ -114,11 +119,13 @@ emaildateformat=%A, %B %d, %Y at %r
|
||||
;
|
||||
; 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
|
||||
; 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.
|
||||
; 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
|
||||
|
||||
@@ -1,17 +1,28 @@
|
||||
;
|
||||
; V6PCI/V12PCI config file for VoiceTronix Hardware
|
||||
; Options
|
||||
; For [general] section
|
||||
;
|
||||
; Options for [general] section
|
||||
;
|
||||
; type = v12pci|v6pci|v4pci
|
||||
; cards = number of cards
|
||||
; indication = 1 ( To use Asterisk indication tones)
|
||||
; ecsuppthres = 0|2048|4096 (none,-24db,-18db only for use with OpenLine4)
|
||||
; dtmfidd = 3000 (Inter Digit Delay timeout for when collecting DTMF tones for dialling from a Station port, in ms)
|
||||
; ast-dtmf-det=1 ( To use Asterisk DTMF detection )
|
||||
; relaxdtmf=1 ( Used with ast-dtmf-det )
|
||||
; break-for-dtmf=no (When a native bridge occurs between 2 vpb channels, it will only break the connection for '#' and '*')
|
||||
; timer_period_ring=4000 (Set the maximum period between received rings, default 4000ms)
|
||||
; 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
|
||||
;
|
||||
; For [interface] section
|
||||
; 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
|
||||
|
||||
@@ -103,9 +103,9 @@ switchtype=national
|
||||
;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
|
||||
; 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
|
||||
@@ -129,58 +129,66 @@ switchtype=national
|
||||
; 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)
|
||||
; 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.
|
||||
; 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
|
||||
; 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
|
||||
; 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)
|
||||
; 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:
|
||||
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
|
||||
; parameters:
|
||||
;defaultozz=0000
|
||||
;defaultcic=303
|
||||
;
|
||||
@@ -197,7 +205,8 @@ 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)
|
||||
; 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
|
||||
@@ -210,12 +219,15 @@ rxwink=300 ; Atlas seems to use long (250ms) winks
|
||||
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
|
||||
; 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
|
||||
; ring = a ring signals the start
|
||||
; polarity = polarity reversal signals the start
|
||||
;
|
||||
;cidstart=ring
|
||||
;
|
||||
@@ -227,12 +239,14 @@ hidecallerid=no
|
||||
;
|
||||
callwaiting=yes
|
||||
;
|
||||
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user)
|
||||
; 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
|
||||
; Whether or not use the caller ID presentation for the outgoing call that the
|
||||
; calling switch is sending.
|
||||
;
|
||||
usecallingpres=yes
|
||||
;
|
||||
@@ -271,31 +285,29 @@ 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.
|
||||
; 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
|
||||
; 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:
|
||||
; 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.
|
||||
; 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.
|
||||
; 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
|
||||
;
|
||||
@@ -309,10 +321,9 @@ echocancelwhenbridged=yes
|
||||
;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.
|
||||
; 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
|
||||
;
|
||||
@@ -321,8 +332,8 @@ echocancelwhenbridged=yes
|
||||
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.
|
||||
; Logical groups can be assigned to allow outgoing rollover. Groups range
|
||||
; from 0 to 63, and multiple groups can be specified.
|
||||
;
|
||||
group=1
|
||||
;
|
||||
@@ -335,19 +346,18 @@ callgroup=1
|
||||
pickupgroup=1
|
||||
|
||||
;
|
||||
; Specify whether the channel should be answered immediately or
|
||||
; if the simple switch should provide dialtone, read digits, etc.
|
||||
; 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).
|
||||
; 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 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
|
||||
;
|
||||
@@ -373,39 +383,36 @@ immediate=no
|
||||
;
|
||||
;busydetect=yes
|
||||
;
|
||||
; If busydetect is enabled, 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
|
||||
; higher the number, more time is needed to hangup a channel, but
|
||||
; lower is probability to get random hangups
|
||||
; 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, is also possible to specify the
|
||||
; cadence of your busy signal. In many countries it is 500mec
|
||||
; on, 500msec off.
|
||||
; Without busypattern specified, we'll accept any regular
|
||||
; sound-silence pattern than 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.
|
||||
; 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
|
||||
; lengh tone and silence (as many countries do), consider defining
|
||||
; the -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
|
||||
; 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
|
||||
; 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
|
||||
;
|
||||
@@ -413,13 +420,13 @@ immediate=no
|
||||
; 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.
|
||||
; 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"
|
||||
; 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.
|
||||
; 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
|
||||
@@ -446,15 +453,15 @@ immediate=no
|
||||
;
|
||||
;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.
|
||||
; 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
|
||||
@@ -465,10 +472,10 @@ immediate=no
|
||||
;
|
||||
;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.
|
||||
; 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[...]]
|
||||
;
|
||||
@@ -479,11 +486,11 @@ immediate=no
|
||||
;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
|
||||
; 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.:
|
||||
; 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
|
||||
@@ -506,9 +513,8 @@ immediate=no
|
||||
;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.
|
||||
; 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
|
||||
@@ -538,9 +544,9 @@ immediate=no
|
||||
;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.
|
||||
; 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
|
||||
|
||||
Reference in New Issue
Block a user