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Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. 

Let's try that again, this time removing trailing whitespace and not leading
whitespace.  I can't believe no one noticed.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
seanbright 2009-05-28 14:39:21 +00:00
parent 7f7cfd42e9
commit a22b4735e5
57 changed files with 1687 additions and 1687 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

22
configs/alsa.conf.sample
View File

@ -39,23 +39,23 @@ extension=s
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The ALSA channel can't accept jitter,
; thus an enabled jitterbuffer on the receive ALSA side will always
; be used if the sending side can create jitter.
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The ALSA channel can't accept jitter,
; thus an enabled jitterbuffer on the receive ALSA side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to 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.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

10
configs/amd.conf.sample
View File

@ -4,15 +4,15 @@
[general]
initial_silence = 2500 ; Maximum silence duration before the greeting.
; If exceeded then MACHINE.
; If exceeded then MACHINE.
greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
after_greeting_silence = 800 ; Silence after detecting a greeting.
; If exceeded then HUMAN
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
; on a HUMAN or MACHINE
; on a HUMAN or MACHINE
min_word_length = 100 ; Minimum duration of Voice to considered as a word
between_words_silence = 50 ; Minimum duration of silence after a word to consider
; the audio what follows as a new word
; the audio what follows as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded then MACHINE
; If exceeded then MACHINE
silence_threshold = 256

180
configs/asterisk.adsi
View File

@ -35,39 +35,39 @@ DISPLAY "empty" IS "asdf"
; Begin soft key definitions
;
KEY "callfwd" IS "CallFwd" OR "Call Forward"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
GOTO "offHook"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
GOTO "offHook"
ENDKEY
KEY "vmail_OH" IS "VMail" OR "Voicemail"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "8500"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "8500"
ENDKEY
KEY "vmail" IS "VMail" OR "Voicemail"
SENDDTMF "8500"
SENDDTMF "8500"
ENDKEY
KEY "backspace" IS "BackSpc" OR "Backspace"
BACKSPACE
BACKSPACE
ENDKEY
KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
SENDDTMF "*70"
SETFLAG "nocallwaiting"
SHOWDISPLAY "cwdisabled" AT 4
TIMERCLEAR
TIMERSTART 1
SENDDTMF "*70"
SETFLAG "nocallwaiting"
SHOWDISPLAY "cwdisabled" AT 4
TIMERCLEAR
TIMERSTART 1
ENDKEY
KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
SENDDTMF "*67"
SETFLAG "nocallwaiting"
SENDDTMF "*67"
SETFLAG "nocallwaiting"
ENDKEY
;
@ -75,85 +75,85 @@ ENDKEY
;
SUB "main" IS
IFEVENT NEARANSWER THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "talkingto" AT 2 NOUPDATE
SHOWDISPLAY "callname" AT 3
SHOWDISPLAY "callnum" AT 4
GOTO "stableCall"
ENDIF
IFEVENT OFFHOOK THEN
CLEAR
CLEARFLAG "nocallwaiting"
CLEARDISPLAY
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail"
SHOWKEYS "cidblock"
SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
GOTO "offHook"
ENDIF
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail_OH"
ENDIF
IFEVENT CALLERID THEN
CLEAR
IFEVENT NEARANSWER THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "talkingto" AT 2 NOUPDATE
SHOWDISPLAY "callname" AT 3
SHOWDISPLAY "callnum" AT 4
GOTO "stableCall"
ENDIF
IFEVENT OFFHOOK THEN
CLEAR
CLEARFLAG "nocallwaiting"
CLEARDISPLAY
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail"
SHOWKEYS "cidblock"
SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
GOTO "offHook"
ENDIF
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail_OH"
ENDIF
IFEVENT CALLERID THEN
CLEAR
; SHOWDISPLAY "titles" AT 1 NOUPDATE
; SHOWDISPLAY "incoming" AT 2 NOUPDATE
SHOWDISPLAY "callname" AT 3 NOUPDATE
SHOWDISPLAY "callnum" AT 4
ENDIF
IFEVENT RING THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "incoming" AT 2
ENDIF
IFEVENT ENDOFRING THEN
SHOWDISPLAY "missedcall" AT 2
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail_OH"
ENDIF
IFEVENT TIMER THEN
CLEAR
SHOWDISPLAY "empty" AT 4
ENDIF
SHOWDISPLAY "callname" AT 3 NOUPDATE
SHOWDISPLAY "callnum" AT 4
ENDIF
IFEVENT RING THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "incoming" AT 2
ENDIF
IFEVENT ENDOFRING THEN
SHOWDISPLAY "missedcall" AT 2
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail_OH"
ENDIF
IFEVENT TIMER THEN
CLEAR
SHOWDISPLAY "empty" AT 4
ENDIF
ENDSUB
SUB "offHook" IS
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "ringing" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 2
GOTO "stableCall"
ENDIF
IFEVENT BUSY THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "busy" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT REORDER THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "reorder" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "ringing" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 2
GOTO "stableCall"
ENDIF
IFEVENT BUSY THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "busy" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT REORDER THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "reorder" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
ENDSUB
SUB "stableCall" IS
IFEVENT REORDER THEN
SHOWDISPLAY "callended" AT 2
ENDIF
IFEVENT REORDER THEN
SHOWDISPLAY "callended" AT 2
ENDIF
ENDSUB

28
configs/chan_dahdi.conf.sample
View File

@ -581,9 +581,9 @@ pickupgroup=1
; Channel variable to be set for all calls from this channel
;setvar=CHANNEL=42
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;
; Specify whether the channel should be answered immediately or if the simple
@ -792,23 +792,23 @@ pickupgroup=1
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The DAHDI channel can't accept jitter,
; thus an enabled jitterbuffer on the receive DAHDI side will always
; be used if the sending side can create jitter.
; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The DAHDI channel can't accept jitter,
; thus an enabled jitterbuffer on the receive DAHDI side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; 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 DAHDI
; 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.
; 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.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

4
configs/cli_aliases.conf.sample
View File

@ -13,8 +13,8 @@ template = friendly ; By default, include friendly aliases
;template = asterisk12 ; Asterisk 1.2 style syntax
;template = asterisk14 ; Asterisk 1.4 style syntax
;template = individual_custom ; see [individual_custom] example below which
; includes a list of aliases from an external
; file
; includes a list of aliases from an external
; file
; Because the Asterisk CLI syntax follows a "module verb argument" syntax,

2
configs/cli_permissions.conf.sample
View File

@ -23,7 +23,7 @@
[general]
default_perm=permit ; To leave asterisk working as normal
; we should set this parameter to 'permit'
; we should set this parameter to 'permit'
;
; Follows the per-users permissions configs.
;

32
configs/console.conf.sample
View File

@ -34,7 +34,7 @@
; The default is "no".
;
;overridecontext = no ; if 'no', the last @ will start the context
; if 'yes' the whole string is an extension.
; if 'yes' the whole string is an extension.
; Default Music on Hold class to use when this channel is placed on hold in
@ -46,23 +46,23 @@
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; Console channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The Console channel can't accept jitter,
; thus an enabled jitterbuffer on the receive Console side will always
; be used if the sending side can create jitter.
; Console channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The Console channel can't accept jitter,
; thus an enabled jitterbuffer on the receive Console side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; 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 Console
; 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.
; 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.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -76,8 +76,8 @@
[default]
input_device = default ; When configuring an input device and output device,
output_device = default ; use the name that you see when you run the "console
; list available" CLI command. If you say "default", the
; system default input and output devices will be used.
; list available" CLI command. If you say "default", the
; system default input and output devices will be used.
autoanswer = no
context = default
extension = s
@ -86,5 +86,5 @@ language = en
overridecontext = no
mohinterpret = default
active = yes ; This option should only be set for one console.
; It means that it is the active console to be
; used from the Asterisk CLI.
; It means that it is the active console to be
; used from the Asterisk CLI.

4
configs/dnsmgr.conf.sample
View File

@ -1,5 +1,5 @@
[general]
;enable=yes ; enable creation of managed DNS lookups
; default is 'no'
; default is 'no'
;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds
; default is 300 (5 minutes)
; default is 300 (5 minutes)

376
configs/extensions.ael.sample
View File

@ -19,28 +19,28 @@
//
globals {
CONSOLE="Console/dsp"; // Console interface for demo
//CONSOLE=DAHDI/1
//CONSOLE=Phone/phone0
IAXINFO=guest; // IAXtel username/password
//IAXINFO="myuser:mypass";
TRUNK="DAHDI/G2"; // Trunk interface
//
// Note the 'G2' in the TRUNK variable above. It specifies which group (defined
// in dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
// the specified group. The four possible options are:
//
// g: select the lowest-numbered non-busy DAHDI channel
// (aka. ascending sequential hunt group).
// G: select the highest-numbered non-busy DAHDI channel
// (aka. descending sequential hunt group).
// r: use a round-robin search, starting at the next highest channel than last
// time (aka. ascending rotary hunt group).
// R: use a round-robin search, starting at the next lowest channel than last
// time (aka. descending rotary hunt group).
//
TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
//TRUNK=IAX2/user:pass@provider
CONSOLE="Console/dsp"; // Console interface for demo
//CONSOLE=DAHDI/1
//CONSOLE=Phone/phone0
IAXINFO=guest; // IAXtel username/password
//IAXINFO="myuser:mypass";
TRUNK="DAHDI/G2"; // Trunk interface
//
// Note the 'G2' in the TRUNK variable above. It specifies which group (defined
// in dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
// the specified group. The four possible options are:
//
// g: select the lowest-numbered non-busy DAHDI channel
// (aka. ascending sequential hunt group).
// G: select the highest-numbered non-busy DAHDI channel
// (aka. descending sequential hunt group).
// r: use a round-robin search, starting at the next highest channel than last
// time (aka. ascending rotary hunt group).
// R: use a round-robin search, starting at the next lowest channel than last
// time (aka. descending rotary hunt group).
//
TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
//TRUNK=IAX2/user:pass@provider
};
//
@ -110,61 +110,61 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
//
//
context ael-dundi-e164-canonical {
//
// List canonical entries here
//
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
//
// List canonical entries here
//
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
};
context ael-dundi-e164-customers {
//
// If you are an ITSP or Reseller, list your customers here.
//
//_12564286000 => Dial(SIP/customer1);
//_12564286001 => Dial(IAX2/customer2);
//
// If you are an ITSP or Reseller, list your customers here.
//
//_12564286000 => Dial(SIP/customer1);
//_12564286001 => Dial(IAX2/customer2);
};
context ael-dundi-e164-via-pstn {
//
// If you are freely delivering calls to the PSTN, list them here
//
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
//
// If you are freely delivering calls to the PSTN, list them here
//
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
};
context ael-dundi-e164-local {
//
// Context to put your dundi IAX2 or SIP user in for
// full access
//
includes {
ael-dundi-e164-canonical;
ael-dundi-e164-customers;
ael-dundi-e164-via-pstn;
};
//
// Context to put your dundi IAX2 or SIP user in for
// full access
//
includes {
ael-dundi-e164-canonical;
ael-dundi-e164-customers;
ael-dundi-e164-via-pstn;
};
};
context ael-dundi-e164-switch {
//
// Just a wrapper for the switch
//
//
// Just a wrapper for the switch
//
switches {
DUNDi/e164;
};
switches {
DUNDi/e164;
};
};
context ael-dundi-e164-lookup {
//
// Locally to lookup, try looking for a local E.164 solution
// then try DUNDi if we don't have one.
//
includes {
ael-dundi-e164-local;
ael-dundi-e164-switch;
};
//
//
// Locally to lookup, try looking for a local E.164 solution
// then try DUNDi if we don't have one.
//
includes {
ael-dundi-e164-local;
ael-dundi-e164-switch;
};
//
};
//
@ -175,8 +175,8 @@ macro ael-dundi-e164(exten) {
//
// ARG1 is the extension to Dial
//
goto ${exten}|1;
return;
goto ${exten}|1;
return;
};
//
@ -186,7 +186,7 @@ return;
// up, please go to www.gnophone.com or www.iaxtel.com
//
context ael-iaxtel700 {
_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
};
//
@ -196,91 +196,91 @@ _91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
// to be on-line or else dialing can be severly delayed.
//
context ael-iaxprovider {
switches {
// IAX2/user:[key]@myserver/mycontext;
};
switches {
// IAX2/user:[key]@myserver/mycontext;
};
};
context ael-trunkint {
//
// International long distance through trunk
//
includes {
ael-dundi-e164-lookup;
};
_9011. => {
&ael-dundi-e164(${EXTEN:4});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
//
// International long distance through trunk
//
includes {
ael-dundi-e164-lookup;
};
_9011. => {
&ael-dundi-e164(${EXTEN:4});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
};
context ael-trunkld {
//
// Long distance context accessed through trunk
//
includes {
ael-dundi-e164-lookup;
};
_91NXXNXXXXXX => {
&ael-dundi-e164(${EXTEN:1});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
//
// Long distance context accessed through trunk
//
includes {
ael-dundi-e164-lookup;
};
_91NXXNXXXXXX => {
&ael-dundi-e164(${EXTEN:1});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
};
context ael-trunklocal {
//
// Local seven-digit dialing accessed through trunk interface
//
_9NXXXXXX => {
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
//
// Local seven-digit dialing accessed through trunk interface
//
_9NXXXXXX => {
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
};
context ael-trunktollfree {
//
// Long distance context accessed through trunk interface
//
//
// Long distance context accessed through trunk interface
//
_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
context ael-international {
//
// Master context for international long distance
//
ignorepat => 9;
includes {
ael-longdistance;
ael-trunkint;
};
//
// Master context for international long distance
//
ignorepat => 9;
includes {
ael-longdistance;
ael-trunkint;
};
};
context ael-longdistance {
//
// Master context for long distance
//
ignorepat => 9;
includes {
ael-local;
ael-trunkld;
};
//
// Master context for long distance
//
ignorepat => 9;
includes {
ael-local;
ael-trunkld;
};
};
context ael-local {
//
// Master context for local, toll-free, and iaxtel calls only
//
ignorepat => 9;
includes {
ael-default;
ael-trunklocal;
ael-iaxtel700;
ael-trunktollfree;
ael-iaxprovider;
};
//
// Master context for local, toll-free, and iaxtel calls only
//
ignorepat => 9;
includes {
ael-default;
ael-trunklocal;
ael-iaxtel700;
ael-trunktollfree;
ael-iaxprovider;
};
};
//
@ -306,69 +306,69 @@ ael-iaxprovider;
macro ael-std-exten-ael( ext , dev ) {
Dial(${dev}/${ext},20);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(${ext},b);
break;
default:
Voicemail(${ext},u);
};
catch a {
VoiceMailMain(${ext});
return;
};
return;
Dial(${dev}/${ext},20);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(${ext},b);
break;
default:
Voicemail(${ext},u);
};
catch a {
VoiceMailMain(${ext});
return;
};
return;
};
context ael-demo {
s => {
Wait(1);
Answer();
Set(TIMEOUT(digit)=5);
Set(TIMEOUT(response)=10);
s => {
Wait(1);
Answer();
Set(TIMEOUT(digit)=5);
Set(TIMEOUT(response)=10);
restart:
Background(demo-congrats);
Background(demo-congrats);
instructions:
for (x=0; ${x} < 3; x=${x} + 1) {
Background(demo-instruct);
WaitExten();
};
};
2 => {
Background(demo-moreinfo);
goto s|instructions;
};
3 => {
Set(LANGUAGE()=fr);
goto s|restart;
};
1000 => {
goto ael-default|s|1;
};
500 => {
Playback(demo-abouttotry);
Dial(IAX2/guest@misery.digium.com/s@default);
Playback(demo-nogo);
goto s|instructions;
};
600 => {
Playback(demo-echotest);
Echo();
Playback(demo-echodone);
goto s|instructions;
};
_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
8500 => {
VoicemailMain();
goto s|instructions;
};
# => {
Playback(demo-thanks);
Hangup();
};
t => goto #|1;
i => Playback(invalid);
for (x=0; ${x} < 3; x=${x} + 1) {
Background(demo-instruct);
WaitExten();
};
};
2 => {
Background(demo-moreinfo);
goto s|instructions;
};
3 => {
Set(LANGUAGE()=fr);
goto s|restart;
};
1000 => {
goto ael-default|s|1;
};
500 => {
Playback(demo-abouttotry);
Dial(IAX2/guest@misery.digium.com/s@default);
Playback(demo-nogo);
goto s|instructions;
};
600 => {
Playback(demo-echotest);
Echo();
Playback(demo-echodone);
goto s|instructions;
};
_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
8500 => {
VoicemailMain();
goto s|instructions;
};
# => {
Playback(demo-thanks);
Hangup();
};
t => goto #|1;
i => Playback(invalid);
};
@ -383,9 +383,9 @@ context ael-default {
// By default we include the demo. In a production system, you
// probably don't want to have the demo there.
includes {
ael-demo;
};
includes {
ael-demo;
};
//
// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
// Note that you must have a [sipprovider] section in sip.conf whereas

10
configs/extensions.conf.sample
View File

@ -430,7 +430,7 @@ exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b)
; If busy, send to voicemail w/ busy announce
; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
@ -459,7 +459,7 @@ exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
@ -521,7 +521,7 @@ exten => 1000,1,Goto(default,s,1)
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
; (but skip if channel is not up)
exten => 1234,n,Gosub(stdexten(1234,${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
@ -640,11 +640,11 @@ include => demo
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Gosub(stdexten(6275,${MARK}))
; assuming ${MARK} is something like DAHDI/2
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(stdexten(6236,${WIL}))
; Ditto for wil
; Ditto for wil
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)

136
configs/extensions.lua.sample
View File

@ -97,103 +97,103 @@ TRUNKMSD = 1
--
function outgoing_local(c, e)
app.dial("DAHDI/1/" .. e, "", "")
app.dial("DAHDI/1/" .. e, "", "")
end
function demo_instruct()
app.background("demo-instruct")
app.waitexten()
app.background("demo-instruct")
app.waitexten()
end
function demo_congrats()
app.background("demo-congrats")
demo_instruct()
app.background("demo-congrats")
demo_instruct()
end
-- Answer the chanel and play the demo sound files
function demo_start(context, exten)
app.wait(1)
app.answer()
app.wait(1)
app.answer()
channel.TIMEOUT("digit"):set(5)
channel.TIMEOUT("response"):set(10)
-- app.set("TIMEOUT(digit)=5")
-- app.set("TIMEOUT(response)=10")
channel.TIMEOUT("digit"):set(5)
channel.TIMEOUT("response"):set(10)
-- app.set("TIMEOUT(digit)=5")
-- app.set("TIMEOUT(response)=10")
demo_congrats(context, exten)
demo_congrats(context, exten)
end
function demo_hangup()
app.playback("demo-thanks")
app.hangup()
app.playback("demo-thanks")
app.hangup()
end
extensions = {
demo = {
s = demo_start;
demo = {
s = demo_start;
["2"] = function()
app.background("demo-moreinfo")
demo_instruct()
end;
["3"] = function ()
channel.LANGUAGE():set("fr") -- set the language to french
demo_congrats()
end;
["2"] = function()
app.background("demo-moreinfo")
demo_instruct()
end;
["3"] = function ()
channel.LANGUAGE():set("fr") -- set the language to french
demo_congrats()
end;
["1000"] = function()
app.goto("default", "s", 1)
end;
["1000"] = function()
app.goto("default", "s", 1)
end;
["1234"] = function()
app.playback("transfer", "skip")
-- do a dial here
end;
["1234"] = function()
app.playback("transfer", "skip")
-- do a dial here
end;
["1235"] = function()
app.voicemail("1234", "u")
end;
["1235"] = function()
app.voicemail("1234", "u")
end;
["1236"] = function()
app.dial("Console/dsp")
app.voicemail(1234, "b")
end;
["1236"] = function()
app.dial("Console/dsp")
app.voicemail(1234, "b")
end;
["#"] = demo_hangup;
t = demo_hangup;
i = function()
app.playback("invalid")
demo_instruct()
end;
["#"] = demo_hangup;
t = demo_hangup;
i = function()
app.playback("invalid")
demo_instruct()
end;
["500"] = function()
app.playback("demo-abouttotry")
app.dial("IAX2/guest@misery.digium.com/s@default")
app.playback("demo-nogo")
demo_instruct()
end;
["500"] = function()
app.playback("demo-abouttotry")
app.dial("IAX2/guest@misery.digium.com/s@default")
app.playback("demo-nogo")
demo_instruct()
end;
["600"] = function()
app.playback("demo-echotest")
app.echo()
app.playback("demo-echodone")
demo_instruct()
end;
["600"] = function()
app.playback("demo-echotest")
app.echo()
app.playback("demo-echodone")
demo_instruct()
end;
["8500"] = function()
app.voicemailmain()
demo_instruct()
end;
["8500"] = function()
app.voicemailmain()
demo_instruct()
end;
};
};
default = {
-- by default, do the demo
include = {"demo"};
};
default = {
-- by default, do the demo
include = {"demo"};
};
["local"] = {
["_NXXXXXX"] = outgoing_local;
};
["local"] = {
["_NXXXXXX"] = outgoing_local;
};
}

42
configs/features.conf.sample
View File

@ -5,52 +5,52 @@
[general]
parkext => 700 ; What extension to dial to park (all parking lots)
parkpos => 701-720 ; What extensions to park calls on. (defafult parking lot)
; These needs to be numeric, as Asterisk starts from the start position
; and increments with one for the next parked call.
; These needs to be numeric, as Asterisk starts from the start position
; and increments with one for the next parked call.
context => parkedcalls ; Which context parked calls are in (default parking lot)
;parkinghints = no ; Add hints priorities automatically for parking slots (default is no).
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
; (default is 45 seconds)
;comebacktoorigin = yes ; Whether to return to the original calling extension upon parking
; timeout or to send the call to context 'parkedcallstimeout' at
; extension 's', priority '1' (default is yes).
; timeout or to send the call to context 'parkedcallstimeout' at
; extension 's', priority '1' (default is yes).
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
;parkedplay = caller ; Who to play the courtesy tone to when picking up a parked call
; one of: parked, caller, both (default is caller)
; one of: parked, caller, both (default is caller)
;parkedcalltransfers = caller ; Enables or disables DTMF based transfers when picking up a parked call.
; one of: callee, caller, both, no (default is no)
; one of: callee, caller, both, no (default is no)
;parkedcallreparking = caller ; Enables or disables DTMF based parking when picking up a parked call.
; one of: callee, caller, both, no (default is no)
; one of: callee, caller, both, no (default is no)
;parkedcallhangup = caller ; Enables or disables DTMF based hangups when picking up a parked call.
; one of: callee, caller, both, no (default is no)
; one of: callee, caller, both, no (default is no)
;parkedcallrecording = caller ; Enables or disables DTMF based one-touch recording when picking up a parked call.
; one of: callee, caller, both, no (default is no)
; one of: callee, caller, both, no (default is no)
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the 'next' free parking space.
; Defaults to 'first' available
; Defaults to 'first' available
;parkedmusicclass=default ; This is the MOH class to use for the parked channel
; as long as the class is not set on the channel directly
; using Set(CHANNEL(musicclass)=whatever) in the dialplan
; as long as the class is not set on the channel directly
; using Set(CHANNEL(musicclass)=whatever) in the dialplan
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
; (default is 3 seconds)
; (default is 3 seconds)
;xfersound = beep ; to indicate an attended transfer is complete
;xferfailsound = beeperr ; to indicate a failed transfer
;pickupexten = *8 ; Configure the pickup extension. (default is *8)
;pickupsound = beep ; to indicate a successful pickup (default: no sound)
;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound)
;featuredigittimeout = 1000 ; Max time (ms) between digits for
; feature activation (default is 1000 ms)
; feature activation (default is 1000 ms)
;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds.
;atxferdropcall = no ; If someone does an attended transfer, then hangs up before the transferred
; caller is connected, then by default, the system will try to call back the
; person that did the transfer. If this is set to "yes", the callback will
; not be attempted and the transfer will just fail.
; caller is connected, then by default, the system will try to call back the
; person that did the transfer. If this is set to "yes", the callback will
; not be attempted and the transfer will just fail.
;atxferloopdelay = 10 ; Number of seconds to sleep between retries (if atxferdropcall = no)
;atxfercallbackretries = 2 ; Number of times to attempt to send the call back to the transferer.
; By default, this is 2.
; By default, this is 2.
; Note that the DTMF features listed below only work when two channels have answered and are bridged together.
; They can not be used while the remote party is ringing or in progress. If you require this feature you can use

8
configs/func_odbc.conf.sample
View File

@ -76,10 +76,10 @@ readsql=${ARG1}
; ODBC_ANTIGF - A blacklist.
[ANTIGF]
dsn=mysql1,mysql2 ; Use mysql1 as the primary handle, but fall back to mysql2
; if mysql1 is down. Supports up to 5 comma-separated
; DSNs. "dsn" may also be specified as "readhandle" and
; "writehandle", if it is important to separate reads and
; writes to different databases.
; if mysql1 is down. Supports up to 5 comma-separated
; DSNs. "dsn" may also be specified as "readhandle" and
; "writehandle", if it is important to separate reads and
; writes to different databases.
readsql=SELECT COUNT(*) FROM exgirlfriends WHERE callerid='${SQL_ESC(${ARG1})}'
syntax=<callerid>
synopsis=Check if a specified callerid is contained in the ex-gf database

6
configs/gtalk.conf.sample
View File

@ -2,7 +2,7 @@
;context=default ;;Context to dump call into
;bindaddr=0.0.0.0 ;;Address to bind to
;allowguest=yes ;;Allow calls from people not in
;;list of peers
;;list of peers
;
;[guest] ;;special account for options on guest account
;disallow=all
@ -11,10 +11,10 @@
;
;[ogorman]
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;;calling or accepting calls from
;disallow=all
;allow=ulaw
;context=default
;connection=asterisk ;;client or component in jabber.conf
;;for the call to leave on.
;;for the call to leave on.
;

26
configs/h323.conf.sample
View File

@ -122,27 +122,27 @@ port = 1720
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; H323 channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The H323 channel can accept jitter,
; thus an enabled jitterbuffer on the receive H323 side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; H323 channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The H323 channel can accept jitter,
; thus an enabled jitterbuffer on the receive H323 side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a H323
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

52
configs/iax.conf.sample
View File

@ -12,9 +12,9 @@
[general]
;bindport=4569 ; bindport and bindaddr may be specified
; ; NOTE: bindport must be specified BEFORE
; bindaddr or may be specified on a specific
; bindaddr if followed by colon and port
; (e.g. bindaddr=192.168.0.1:4569)
; bindaddr or may be specified on a specific
; bindaddr if followed by colon and port
; (e.g. bindaddr=192.168.0.1:4569)
;bindaddr=192.168.0.1 ; more than once to bind to multiple
; ; addresses, but the first will be the
; ; default
@ -284,29 +284,29 @@ autokill=yes
;allowfwdownload=yes
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a IAX2 peer registers successfully,
; the ip address, the origination port, the registration period,
; and the username of the peer will be set to database via realtime.
; If not present, defaults to 'yes'.
; If set to yes, when a IAX2 peer registers successfully,
; the ip address, the origination port, the registration period,
; and the username of the peer will be set to database via realtime.
; If not present, defaults to 'yes'.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again.
; If set to an integer, friends expire within this number of
; seconds instead of the registration interval.
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again.
; If set to an integer, friends expire within this number of
; seconds instead of the registration interval.
;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration
; has expired based on its registration interval, used the stored
; address information regardless. (yes|no)
; has expired based on its registration interval, used the stored
; address information regardless. (yes|no)
;parkinglot=edvina ; Default parkinglot for IAX peers and users
; This can also be configured per device
; Parkinglots are defined in features.conf
; This can also be configured per device
; Parkinglots are defined in features.conf
; Guest sections for unauthenticated connection attempts. Just specify an
; empty secret, or provide no secret section.
@ -377,13 +377,13 @@ inkeys=freeworlddialup
;auth=md5,plaintext,rsa
;secret=markpasswd
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too
;transfer=no ; Disable IAX native transfer
;transfer=mediaonly ; When doing IAX native transfers, transfer
; only media stream
; only media stream
;jitterbuffer=yes ; Override global setting an enable jitter buffer
; ; for this user
;maxauthreq=10 ; Set maximum number of outstanding AUTHREQs waiting for replies. Any further authentication attempts will be blocked
@ -414,12 +414,12 @@ host=216.207.245.47
;mask=255.255.255.255
;qualify=yes ; Make sure this peer is alive
;qualifysmoothing = yes ; use an average of the last two PONG
; results to reduce falsely detected LAGGED hosts
; Default: Off
; results to reduce falsely detected LAGGED hosts
; Default: Off
;qualifyfreqok = 60000 ; how frequently to ping the peer when
; everything seems to be ok, in milliseconds
; everything seems to be ok, in milliseconds
;qualifyfreqnotok = 10000 ; how frequently to ping the peer when it's
; either LAGGED or UNAVAILABLE, in milliseconds
; either LAGGED or UNAVAILABLE, in milliseconds
;jitterbuffer=no ; Turn off jitter buffer for this peer
;
;encryption=yes ; Enable IAX2 encryption. The default is no.

10
configs/jabber.conf.sample
View File

@ -1,14 +1,14 @@
[general]
;debug=yes ;;Turn on debugging by default.
;autoprune=yes ;;Auto remove users from buddy list. Depending on your
;;setup (ie, using your personal Gtalk account for a test)
;;you might lose your contacts list. Default is 'no'.
;;setup (ie, using your personal Gtalk account for a test)
;;you might lose your contacts list. Default is 'no'.
;autoregister=yes ;;Auto register users from buddy list.
;[asterisk] ;;label
;type=client ;;Client or Component connection
;serverhost=astjab.org ;;Route to server for example,
;; talk.google.com
;; talk.google.com
;username=asterisk@astjab.org/asterisk ;;Username with optional resource.
;secret=blah ;;Password
;priority=1 ;;Resource priority
@ -17,7 +17,7 @@
;usesasl=yes ;;Use sasl or not
;buddy=mogorman@astjab.org ;;Manual addition of buddy to list.
;status=available ;;One of: chat, available, away,
;; xaway, or dnd
;; xaway, or dnd
;statusmessage="I am available" ;;Have custom status message for
;;Asterisk.
;;Asterisk.
;timeout=100 ;;Timeout on the message stack.

6
configs/jingle.conf.sample
View File

@ -2,7 +2,7 @@
;context=default ;;Context to dump call into
;bindaddr=0.0.0.0 ;;Address to bind to
;allowguest=yes ;;Allow calls from people not in
;;list of peers
;;list of peers
;
;[guest] ;;special account for options on guest account
;disallow=all
@ -11,10 +11,10 @@
;
;[ogorman]
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;;calling or accepting calls from
;disallow=all
;allow=ulaw
;context=default
;connection=asterisk ;;client or component in jabber.conf
;;for the call to leave on.
;;for the call to leave on.
;

6
configs/manager.conf.sample
View File

@ -44,8 +44,8 @@ bindaddr = 0.0.0.0
;tlsbindaddr=0.0.0.0 ; address to bind to, default to bindaddr
;tlscertfile=/tmp/asterisk.pem ; path to the certificate.
;tlsprivatekey=/tmp/private.pem ; path to the private key, if no private given,
; if no tlsprivatekey is given, default is to search
; tlscertfile for private key.
; if no tlsprivatekey is given, default is to search
; tlscertfile for private key.
;tlscipher=<cipher string> ; string specifying which SSL ciphers to use or not use
;
;allowmultiplelogin = yes ; IF set to no, rejects manager logins that are already in use.
@ -58,7 +58,7 @@ bindaddr = 0.0.0.0
;timestampevents = yes
; debug = on ; enable some debugging info in AMI messages (default off).
; Also accessible through the "manager debug" CLI command.
; Also accessible through the "manager debug" CLI command.
;[mark]
;secret = mysecret
;deny=0.0.0.0/0.0.0.0

14
configs/meetme.conf.sample
View File

@ -5,13 +5,13 @@
[general]
;audiobuffers=32 ; The number of 20ms audio buffers to be used
; when feeding audio frames from non-DAHDI channels
; into the conference; larger numbers will allow
; for the conference to 'de-jitter' audio that arrives
; at different timing than the conference's timing
; source, but can also allow for latency in hearing
; the audio from the speaker. Minimum value is 2,
; maximum value is 32.
; when feeding audio frames from non-DAHDI channels
; into the conference; larger numbers will allow
; for the conference to 'de-jitter' audio that arrives
; at different timing than the conference's timing
; source, but can also allow for latency in hearing
; the audio from the speaker. Minimum value is 2,
; maximum value is 32.
;
; Conferences may be scheduled from realtime?
;schedule=yes

32
configs/mgcp.conf.sample
View File

@ -13,27 +13,27 @@
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; MGCP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The MGCP channel can accept jitter,
; thus an enabled jitterbuffer on the receive MGCP side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; MGCP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The MGCP channel can accept jitter,
; thus an enabled jitterbuffer on the receive MGCP side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a MGCP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a MGCP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to 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.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -79,7 +79,7 @@
;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.
; '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
@ -87,11 +87,11 @@
;[192.168.1.20]
;accountcode = 1000 ; record this in cdr as account identification for billing
;amaflags = billing ; record this in cdr as flagged for 'billing',
; 'documentation', or 'omit'
; 'documentation', or 'omit'
;context = local
;host = 192.168.1.20
;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*'
; another common format is '*'
; another common format is '*'
;callerid = "Duane Cox" <123> ; now lets setup line 1 using per endpoint configuration...
;callwaiting = no
;callreturn = yes

2
configs/minivm.conf.sample
View File

@ -144,7 +144,7 @@ military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
; locale = <locale> ; Locale for LC_TIME - to get weekdays in local language
; ; See your O/S documentation for proper settings for setlocale()
; templatefile = <filename> ; File name (relative to Asterisk configuration directory,
; or absolute
; or absolute
; messagebody = Format ; Message body definition with variables
;
[template-sv_SE_email]

24
configs/misdn.conf.sample
View File

@ -111,26 +111,26 @@ crypt_keys=test,muh
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

4
configs/musiconhold.conf.sample
View File

@ -3,8 +3,8 @@
;
[general]
;cachertclasses=yes ; use 1 instance of moh class for all users who are using it,
; decrease consumable cpu cycles and memory
; disabled by default
; decrease consumable cpu cycles and memory
; disabled by default
; valid mode options:

182
configs/oss.conf.sample
View File

@ -3,75 +3,75 @@
;
[general]
; General config options, with default values shown.
; You should use one section per device, with [general] being used
; for the first device and also as a template for other devices.
;
; All but 'debug' can go also in the device-specific sections.
;
; debug = 0x0 ; misc debug flags, default is 0
; General config options, with default values shown.
; You should use one section per device, with [general] being used
; for the first device and also as a template for other devices.
;
; All but 'debug' can go also in the device-specific sections.
;
; debug = 0x0 ; misc debug flags, default is 0
; Set the device to use for I/O
; device = /dev/dsp
; Set the device to use for I/O
; device = /dev/dsp
; Optional mixer command to run upon startup (e.g. to set
; volume levels, mutes, etc.
; mixer =
; Optional mixer command to run upon startup (e.g. to set
; volume levels, mutes, etc.
; mixer =
; Software mic volume booster (or attenuator), useful for sound
; cards or microphones with poor sensitivity. The volume level
; is in dB, ranging from -20.0 to +20.0
; boost = n ; mic volume boost in dB
; Software mic volume booster (or attenuator), useful for sound
; cards or microphones with poor sensitivity. The volume level
; is in dB, ranging from -20.0 to +20.0
; boost = n ; mic volume boost in dB
; Set the callerid for outgoing calls
; callerid = John Doe <555-1234>
; Set the callerid for outgoing calls
; callerid = John Doe <555-1234>
; autoanswer = no ; no autoanswer on call
; autohangup = yes ; hangup when other party closes
; extension = s ; default extension to call
; context = default ; default context for outgoing calls
; language = "" ; default language
; autoanswer = no ; no autoanswer on call
; autohangup = yes ; hangup when other party closes
; extension = s ; default extension to call
; context = default ; default context for outgoing calls
; language = "" ; default language
; If you set overridecontext to 'yes', then the whole dial string
; will be interpreted as an extension, which is extremely useful
; to dial SIP, IAX and other extensions which use the '@' character.
; The default is 'no' just for backward compatibility, but the
; suggestion is to change it.
; overridecontext = no ; if 'no', the last @ will start the context
; if 'yes' the whole string is an extension.
; If you set overridecontext to 'yes', then the whole dial string
; will be interpreted as an extension, which is extremely useful
; to dial SIP, IAX and other extensions which use the '@' character.
; The default is 'no' just for backward compatibility, but the
; suggestion is to change it.
; overridecontext = no ; if 'no', the last @ will start the context
; if 'yes' the whole string is an extension.
; low level device parameters in case you have problems with the
; device driver on your operating system. You should not touch these
; unless you know what you are doing.
; queuesize = 10 ; frames in device driver
; frags = 8 ; argument to SETFRAGMENT
; low level device parameters in case you have problems with the
; device driver on your operating system. You should not touch these
; unless you know what you are doing.
; queuesize = 10 ; frames in device driver
; frags = 8 ; argument to SETFRAGMENT
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; OSS channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The OSS channel can't accept jitter,
; thus an enabled jitterbuffer on the receive OSS side will always
; be used if the sending side can create jitter.
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; OSS channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The OSS channel can't accept jitter,
; thus an enabled jitterbuffer on the receive OSS side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
; 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.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
; below is an entry for a second console channel
; [card1]
; device = /dev/dsp1 ; alternate device
; device = /dev/dsp1 ; alternate device
; Below are the settings to support video. You can include them
; in your general configuration as [general](+,video)
@ -79,26 +79,26 @@
; Section names used here are only examples.
[my_video](!) ; you can just include in your config
videodevice = /dev/video0 ; uses your V4L webcam as video source
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
videodevice = /dev/video0 ; uses your V4L webcam as video source
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
; video_size is the geometry used by the encoder.
; Depending on the codec your choice is restricted.
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
; video_size is the geometry used by the encoder.
; Depending on the codec your choice is restricted.
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
; You can also set the geometry used for the camera, local display and remote display.
; The local window is on the right, the remote window is on the left.
; Right clicking with the mouse on a video window increases the size,
; center-clicking reduces the size.
camera_size = cif
remote_size = cif
local_size = qcif
; You can also set the geometry used for the camera, local display and remote display.
; The local window is on the right, the remote window is on the left.
; Right clicking with the mouse on a video window increases the size,
; center-clicking reduces the size.
camera_size = cif
remote_size = cif
local_size = qcif
bitrate = 60000 ; rate told to ffmpeg.
fps = 5 ; frames per second from the source.
; qmin = 3 ; quantizer value passed to the encoder.
bitrate = 60000 ; rate told to ffmpeg.
fps = 5 ; frames per second from the source.
; qmin = 3 ; quantizer value passed to the encoder.
; The keypad is made of an image (in any format supported by SDL_image)
; and some configuration entries indicating the location and function of buttons.
@ -115,30 +115,30 @@ fps = 5 ; frames per second from the source.
; diameter of the ellipse.
;
[my_skin](!)
keypad = /tmp/keypad.jpg
region = 1 rect 19 18 67 18 28
region = 2 rect 84 18 133 18 28
region = 3 rect 152 18 201 18 28
region = 4 rect 19 60 67 60 28
region = 5 rect 84 60 133 60 28
region = 6 rect 152 60 201 60 28
region = 7 rect 19 103 67 103 28
region = 8 rect 84 103 133 103 28
region = 9 rect 152 103 201 103 28
region = * rect 19 146 67 146 28
region = 0 rect 84 146 133 146 28
region = # rect 152 146 201 146 28
region = pickup rect 229 15 267 15 40
region = hangup rect 230 66 270 64 40
region = mute circle 232 141 264 141 33
region = sendvideo circle 235 185 266 185 33
region = autoanswer rect 228 212 275 212 50
keypad = /tmp/keypad.jpg
region = 1 rect 19 18 67 18 28
region = 2 rect 84 18 133 18 28
region = 3 rect 152 18 201 18 28
region = 4 rect 19 60 67 60 28
region = 5 rect 84 60 133 60 28
region = 6 rect 152 60 201 60 28
region = 7 rect 19 103 67 103 28
region = 8 rect 84 103 133 103 28
region = 9 rect 152 103 201 103 28
region = * rect 19 146 67 146 28
region = 0 rect 84 146 133 146 28
region = # rect 152 146 201 146 28
region = pickup rect 229 15 267 15 40
region = hangup rect 230 66 270 64 40
region = mute circle 232 141 264 141 33
region = sendvideo circle 235 185 266 185 33
region = autoanswer rect 228 212 275 212 50
; another skin with entries for the keypad and a small font
; to write to the message boards in the skin.
[skin2](!)
keypad = /tmp/kpad2.jpg
keypad_font = /tmp/font.png
keypad = /tmp/kpad2.jpg
keypad_font = /tmp/font.png
; to add video support, uncomment this and remember to install
; the keypad and keypad_font files to the right place

10
configs/phoneprov.conf.sample
View File

@ -6,8 +6,8 @@
;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address.
;serveriface=eth0 ; Same as above, except an ethernet interface.
; Useful for when the interface uses DHCP and the asterisk http
; server listens on a different IP than chan_sip.
; Useful for when the interface uses DHCP and the asterisk http
; server listens on a different IP than chan_sip.
;serverport=5060 ; Override port to send to the phone to use as server port.
default_profile=polycom ; The default profile to use if none specified in users.conf
@ -43,10 +43,10 @@ default_profile=polycom ; The default profile to use if none specified in users.
[polycom]
staticdir => configs/ ; Sub directory of AST_DATA_DIR/phoneprov that static files reside
; in. This allows a request to /phoneprov/sip.cfg to pull the file
; from /phoneprov/configs/sip.cfg
; in. This allows a request to /phoneprov/sip.cfg to pull the file
; from /phoneprov/configs/sip.cfg
mime_type => text/xml ; Default mime type to use if one isn't specified or the
; extension isn't recognized
; extension isn't recognized
static_file => bootrom.ld,application/octet-stream ; Static files the phone will download
static_file => bootrom.ver,plain/text ; static_file => filename,mime-type
static_file => sip.ld,application/octet-stream

20
configs/queues.conf.sample
View File

@ -300,23 +300,23 @@ shared_lastcall=no
;
; queue-thankyou=
;
; ("You are now first in line.")
; ("You are now first in line.")
;queue-youarenext = queue-youarenext
; ("There are")
; ("There are")
;queue-thereare = queue-thereare
; ("calls waiting.")
; ("calls waiting.")
;queue-callswaiting = queue-callswaiting
; ("The current est. holdtime is")
; ("The current est. holdtime is")
;queue-holdtime = queue-holdtime
; ("minutes.")
; ("minutes.")
;queue-minutes = queue-minutes
; ("seconds.")
; ("seconds.")
;queue-seconds = queue-seconds
; ("Thank you for your patience.")
; ("Thank you for your patience.")
;queue-thankyou = queue-thankyou
; ("Hold time")
; ("Hold time")
;queue-reporthold = queue-reporthold
; ("All reps busy / wait for next")
; ("All reps busy / wait for next")
;periodic-announce = queue-periodic-announce
;
; A set of periodic announcements can be defined by separating
@ -501,5 +501,5 @@ shared_lastcall=no
;
;member => Agent/@1 ; Any agent in group 1
;member => Agent/:1,1 ; Any agent in group 1, wait for first
; available, but consider with penalty
; available, but consider with penalty

10
configs/res_odbc.conf.sample
View File

@ -49,11 +49,11 @@ pre-connect => yes
sanitysql => select count(*) from systables
; forcecommit => no ; Default to committing uncommitted transactions?
; isolation => read_committed ; Isolation level; supported levels are:
; read_uncommitted, read_committed, repeatable_read,
; serializable. Note that not all databases support
; all isolation levels (e.g. Postgres only supports
; repeatable_read and serializable). See database
; documentation for further information.
; read_uncommitted, read_committed, repeatable_read,
; serializable. Note that not all databases support
; all isolation levels (e.g. Postgres only supports
; repeatable_read and serializable). See database
; documentation for further information.
;
; Many databases have a default of '\' to escape special characters. MS SQL
; Server does not.

16
configs/rpt.conf.sample
View File

@ -28,13 +28,13 @@
;funcchar = * ; function lead-in character (defaults to '*')
;endchar = # ; command mode end character (defaults to '#')
;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when
; normal patch in use
; normal patch in use
;hangtime=1000 ; squelch tail hang time (in ms) (optional)
;totime=100000 ; transmit time-out time (in ms) (optional)
;idtime=30000 ; id interval time (in ms) (optional)
;politeid=30000 ; time in milliseconds before ID timer
; expires to try and ID in the tail.
; (optional, default is 30000).
; expires to try and ID in the tail.
; (optional, default is 30000).
;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none
;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none
@ -69,13 +69,13 @@
;funcchar = * ; function lead-in character (defaults to '*')
;endchar = # ; command mode end character (defaults to '#')
;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when
; normal patch in use
; normal patch in use
;hangtime=1000 ; squelch tail hang time (in ms) (optional)
;totime=100000 ; transmit time-out time (in ms) (optional)
;idtime=30000 ; id interval time (in ms) (optional)
;politeid=30000 ; time in milliseconds before ID timer
; expires to try and ID in the tail.
; (optional, default is 30000).
; expires to try and ID in the tail.
; (optional, default is 30000).
;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none
;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none
@ -87,8 +87,8 @@
;txchannel = DAHDI/6 ; Tx audio/signalling channel
;functions = functions-remote
;remote = ft897 ; Set remote=y for dumb remote or
; remote=ft897 for Yaesu FT-897 or
; remote=rbi for Doug Hall RBI1
; remote=ft897 for Yaesu FT-897 or
; remote=rbi for Doug Hall RBI1
;iobase = 0x378 ; Specify IO port for parallel port (optional)
;[functions-repeater]

2
configs/rtp.conf.sample
View File

@ -19,7 +19,7 @@ rtpend=20000
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that
; do not come from the source of the RTP stream. This option is

202
configs/say.conf.sample
View File

@ -4,8 +4,8 @@
[general]
mode=old ; method for playing numbers and dates
; old - using asterisk core function
; new - using this configuration file
; old - using asterisk core function
; new - using this configuration file
; The new language routines produce strings of the form
; prefix:[format:]data
@ -75,126 +75,126 @@ mode=old ; method for playing numbers and dates
; language-independent
[digit-base](!) ; base rule for digit strings
; XXX incomplete yet
_digit:[0-9] => digits/${SAY}
_digit:[-] => letters/dash
_digit:[*] => letters/star
_digit:[@] => letters/at
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
; XXX incomplete yet
_digit:[0-9] => digits/${SAY}
_digit:[-] => letters/dash
_digit:[*] => letters/star
_digit:[@] => letters/at
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
[date-base](!) ; base rules for dates and times
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. => num:${SAY:0:4} ; year, 19xx
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
_date:[de]:. => num:${SAY:6:2} ; day of month
_date:[hH]:. => num:${SAY:8:2} ; hour
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[M]:. => num:${SAY:10:2} ; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
_date:[pP]:. => digits/p-m ; am pm
_date:[S]:. => num:${SAY:13:2} ; seconds
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. => num:${SAY:0:4} ; year, 19xx
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
_date:[de]:. => num:${SAY:6:2} ; day of month
_date:[hH]:. => num:${SAY:8:2} ; hour
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[M]:. => num:${SAY:10:2} ; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
_date:[pP]:. => digits/p-m ; am pm
_date:[S]:. => num:${SAY:13:2} ; seconds
[en-base](!)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
; enumeration
_e[n]um:X => digits/h-${SAY}
_e[n]um:1X => digits/h-${SAY}
_e[n]um:[2-9]0 => digits/h-${SAY}
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
; enumeration
_e[n]um:X => digits/h-${SAY}
_e[n]um:1X => digits/h-${SAY}
_e[n]um:[2-9]0 => digits/h-${SAY}
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
[it](digit-base,date-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
_[n]um:1XXX => digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[en](en-base,date-base,digit-base)
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[de](date-base,digit-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[hu](digit-base,date-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1[1-9] => digits/10en, digits/${SAY:1}
_[n]um:2[1-9] => digits/20on, digits/${SAY:1}
_[n]um:[1-9]0 => digits/${SAY}
_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1[1-9] => digits/10en, digits/${SAY:1}
_[n]um:2[1-9] => digits/20on, digits/${SAY:1}
_[n]um:[1-9]0 => digits/${SAY}
_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
_datetime::. => date:YBdA k 'ora' M 'perc':${SAY}
_date::. => date:YBdA:${SAY}
_time::. => date:k 'ora' M 'perc':${SAY}
_datetime::. => date:YBdA k 'ora' M 'perc':${SAY}
_date::. => date:YBdA:${SAY}
_time::. => date:k 'ora' M 'perc':${SAY}

536
configs/sip.conf.sample
View File

@ -88,18 +88,18 @@
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
;match_auth_username=yes ; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
; 'username' field from the authentication line
; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
; Default is enabled
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;
; Note that the TCP and TLS support for chan_sip is currently considered
@ -109,20 +109,20 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0
;
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
; default is to look for "asterisk.pem" in current directory
; default is to look for "asterisk.pem" in current directory
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.
;tlscafile=</path/to/certificate>
; If the server your connecting to uses a self signed certificate
@ -146,20 +146,20 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
@ -173,32 +173,32 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;cos_text=3 ; Sets 802.1p priority for RTP text packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
; the From: header as the "name" portion. Also fill the
; "user" portion of the URI in the From: header with this
; value if no fromuser is set
; Default: empty
; the From: header as the "name" portion. Also fill the
; "user" portion of the URI in the From: header with this
; value if no fromuser is set
; Default: empty
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
; Message-Account in the MWI notify message
; defaults to "asterisk"
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
@ -220,83 +220,83 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;mohsuggest=default
;
;parkinglot=plaza ; Sets the default parking lot for call parking
; This may also be set for individual users/peers
; Parkinglots are configured in features.conf
; This may also be set for individual users/peers
; Parkinglots are configured in features.conf
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;sendrpid = rpid ; Use the "Remote-Party-ID" header
; to send the identity of the remote party
; This is identical to sendrpid=yes
; to send the identity of the remote party
; This is identical to sendrpid=yes
;sendrpid = pai ; Use the "P-Asserted-Identity" header
; to send the identity of the remote party
; to send the identity of the remote party
;rpid_update = no ; In certain cases, the only method by which a connected line
; change may be immediately transmitted is with a SIP UPDATE request.
; If communicating with another Asterisk server, and you wish to be able
; transmit such UPDATE messages to it, then you must enable this option.
; Otherwise, we will have to wait until we can send a reinvite to
; transmit the information.
; change may be immediately transmitted is with a SIP UPDATE request.
; If communicating with another Asterisk server, and you wish to be able
; transmit such UPDATE messages to it, then you must enable this option.
; Otherwise, we will have to wait until we can send a reinvite to
; transmit the information.
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX ; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
; This field MUST NOT contain spaces
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
; Other options:
; info : SIP INFO messages (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
;
;videosupport=yes ; Turn on support for SIP video. You need to turn this
; on in this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can't enable it for
; one peer only without enabling in the general section.
; If you set videosupport to "always", then RTP ports will
; always be set up for video, even on clients that don't
; support it. This assists callfile-derived calls and
; certain transferred calls to use always use video when
; available. [yes|NO|always]
; on in this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can't enable it for
; one peer only without enabling in the general section.
; If you set videosupport to "always", then RTP ports will
; always be set up for video, even on clients that don't
; support it. This assists callfile-derived calls and
; certain transferred calls to use always use video when
; available. [yes|NO|always]
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
; Videosupport and maxcallbitrate is settable
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
; performs events (e.g. hold)
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
; authenticate with Asterisk. Peerstatus will be "rejected".
; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.
; for any reason, always reject with an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
@ -304,18 +304,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
; as any IP address used for staticly defined
; hosts. This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.
; as any IP address used for staticly defined
; hosts. This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.
;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
; register their phones.
; register their phones.
;engine=asterisk ; RTP engine to use when communicating with the device
@ -332,9 +332,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;regcontext=sipregistrations
;regextenonqualify=yes ; Default "no"
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
@ -342,13 +342,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
@ -356,15 +356,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; The settings are settable in the global section as well as per device
;
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
; on the audio channel
; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)
; (default is off - zero)
;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
@ -403,11 +403,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
; the moment the channel loads this configuration
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG logging channel
; SIP history is output to the DEBUG logging channel
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
@ -430,30 +430,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = no ; Control whether subscriptions already INUSE get sent
; RINGING when another call is sent (default: yes)
; RINGING when another call is sent (default: yes)
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
;notifycid = yes ; Control whether caller ID information is sent along with
; dialog-info+xml notifications (supported by snom phones).
; Note that this feature will only work properly when the
; incoming call is using the same extension and context that
; is being used as the hint for the called extension. This means
; that it won't work when using subscribecontext for your sip
; user or peer (if subscribecontext is different than context).
; This is also limited to a single caller, meaning that if an
; extension is ringing because multiple calls are incoming,
; only one will be used as the source of caller ID. Specify
; 'ignore-context' to ignore the called context when looking
; for the caller's channel. The default value is 'no.' Setting
; notifycid to 'ignore-context' also causes call-pickups attempted
; via SNOM's NOTIFY mechanism to set the context for the call pickup
; to PICKUPMARK.
; dialog-info+xml notifications (supported by snom phones).
; Note that this feature will only work properly when the
; incoming call is using the same extension and context that
; is being used as the hint for the called extension. This means
; that it won't work when using subscribecontext for your sip
; user or peer (if subscribecontext is different than context).
; This is also limited to a single caller, meaning that if an
; extension is ringing because multiple calls are incoming,
; only one will be used as the source of caller ID. Specify
; 'ignore-context' to ignore the called context when looking
; for the caller's channel. The default value is 'no.' Setting
; notifycid to 'ignore-context' also causes call-pickups attempted
; via SNOM's NOTIFY mechanism to set the context for the call pickup
; to PICKUPMARK.
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
; device too.
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
@ -536,9 +536,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones.
@ -645,43 +645,43 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;
;canreinvite=yes ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; stay in the audio path, you may want to turn this off.
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; stay in the audio path, you may want to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).
; This setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if canreinvite is enabled when
; the device is actually behind NAT.
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if canreinvite is enabled when
; the device is actually behind NAT.
;canreinvite=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).
;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
; 'canreinvite=update,nonat'. It implies 'yes'.
; instead of INVITE. This can be combined with 'nonat', as
; 'canreinvite=update,nonat'. It implies 'yes'.
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
; number in SDP packets and will only modify the SDP
; session if the version number changes. This option will
; force asterisk to ignore the SDP session version number
; and treat all SDP data as new data. This is required
; for devices that send us non standard SDP packets
; (observed with Microsoft OCS). By default this option is
; off.
; number in SDP packets and will only modify the SDP
; session if the version number changes. This option will
; force asterisk to ignore the SDP session version number
; and treat all SDP data as new data. This is required
; for devices that send us non standard SDP packets
; (observed with Microsoft OCS). By default this option is
; off.
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
@ -689,38 +689,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; source code.
;
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
;rtsavesysname=yes ; Save systemname in realtime database at registration
; Default= no
; Default= no
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime.
; If not present, defaults to 'yes'. Note: realtime peers will
; probably not function across reloads in the way that you expect, if
; you turn this option off.
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime.
; If not present, defaults to 'yes'. Note: realtime peers will
; probably not function across reloads in the way that you expect, if
; you turn this option off.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again. If set
; to an integer, friends expire within this number of seconds
; instead of the registration interval.
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again. If set
; to an integer, friends expire within this number of seconds
; instead of the registration interval.
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
; For non-realtime peers, when their registration expires, the
; information will _not_ be removed from memory or the Asterisk database
; if you attempt to place a call to the peer, the existing information
; will be used in spite of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer
; is still in memory (due to caching or other reasons), the
; information will not be removed from realtime storage
;
; For non-realtime peers, when their registration expires, the
; information will _not_ be removed from memory or the Asterisk database
; if you attempt to place a call to the peer, the existing information
; will be used in spite of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer
; is still in memory (due to caching or other reasons), the
; information will not be removed from realtime storage
;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
@ -744,45 +744,45 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; allowexternaldomains=no
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
; You can have several "domain" settings
; You can have several "domain" settings
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
; Default is yes
; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
; name and local IP to domain list.
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -919,7 +919,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;busylevel=2 ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
;port=80 ; The port number we want to connect to on the remote side
; Also used as "defaultport" in combination with "defaultip" settings
; Also used as "defaultport" in combination with "defaultip" settings
;--- sample definition for a provider
;[provider1]
@ -940,30 +940,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; the templates uncommented as they will not harm:
[basic-options](!) ; a template
dtmfmode=rfc2833
context=from-office
type=friend
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
nat=yes
canreinvite=no
host=dynamic
nat=yes
canreinvite=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
nat=no
canreinvite=yes
nat=no
canreinvite=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
disallow=all
allow=ulaw
; and finally instantiate a few phones
;
@ -982,31 +982,31 @@ allow=ulaw
;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a "friend"
; so there's currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a "friend"
; so there's currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information
; See README.callingpres for more information
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
@ -1035,10 +1035,10 @@ allow=ulaw
;defaultip=192.168.0.59 ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
; subscribes for mailbox notification
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
@ -1051,7 +1051,7 @@ allow=ulaw
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;defaultuser=polly ; Username to use in INVITE until peer registers
;defaultip=192.168.40.123
; Normally you do NOT need to set this parameter
; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don't work properly with "never"
@ -1062,16 +1062,16 @@ allow=ulaw
;secret=blah
;host=dynamic
;insecure=port ; Allow matching of peer by IP address without
; matching port number
; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it's 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
; Helps with NAT session
; qualify=yes uses default value
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
;
; Call group and Pickup group should be in the range from 0 to 63
;
@ -1086,30 +1086,30 @@ allow=ulaw
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;defaultuser=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on or DTMF reception will work improperly.
; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.

52
configs/skinny.conf.sample
View File

@ -5,13 +5,13 @@
bindaddr=0.0.0.0 ; Address to bind to
bindport=2000 ; Port to bind to, default tcp/2000
dateformat=M-D-Y ; M,D,Y in any order (6 chars max)
; "A" may also be used, but it must be at the end.
; Use M for month, D for day, Y for year, A for 12-hour time.
; "A" may also be used, but it must be at the end.
; Use M for month, D for day, Y for year, A for 12-hour time.
keepalive=120
;vmexten=8500 ; Systemwide voicemailmain pilot number
; It must be in the same context as the calling
; device/line
; It must be in the same context as the calling
; device/line
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given line which registers or unregisters with
@ -38,27 +38,27 @@ keepalive=120
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; skinny channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The skinny channel can accept
; jitter, thus a jitterbuffer on the receive skinny side will be
; used only if it is forced and enabled.
; skinny channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The skinny channel can accept
; jitter, thus a jitterbuffer on the receive skinny side will be
; used only if it is forced and enabled.
;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a skinny
; channel. Defaults to "no".
; channel. Defaults to "no".
;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
;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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; 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
; skinny channel. Two implementations are currently available
; - "fixed" (with size always equals to jbmaxsize)
; - "adaptive" (with variable size, actually the new jb of IAX2).
; Defaults to fixed.
; skinny channel. Two implementations are currently available
; - "fixed" (with size always equals to jbmaxsize)
; - "adaptive" (with variable size, actually the new jb of IAX2).
; Defaults to fixed.
;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -94,7 +94,7 @@ keepalive=120
;regexten=100
;context=inbound
;linelabel="Support Line" ; Displays next to the line
; button on 7940's and 7960s
; button on 7940's and 7960s
;[110]
;callerid="John Chambers" <408-526-4000>
;context=did
@ -110,21 +110,21 @@ keepalive=120
;callerid="George W. Bush" <202-456-1414>
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;mailbox=500
;callwaiting=yes
;transfer=yes
;threewaycalling=yes
;context=default
;mohinterpret=default ; This option specifies a default music on hold class to
; use when put on hold if the channel's moh class was not
; explicitly set with Set(CHANNEL(musicclass)=whatever) and
; the peer channel did not suggest a class to use.
; use when put on hold if the channel's moh class was not
; explicitly set with Set(CHANNEL(musicclass)=whatever) and
; the peer channel did not suggest a class to use.
;mohsuggest=default ; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
[devices]

98
configs/sla.conf.sample
View File

@ -8,10 +8,10 @@
[general]
;attemptcallerid=no ; Attempt CallerID handling. The default value for this
; is "no" because CallerID handling with an SLA setup is
; known to not work properly in some situations. However,
; feel free to enable it if you would like. If you do, and
; you find problems, please do not report them.
; is "no" because CallerID handling with an SLA setup is
; known to not work properly in some situations. However,
; feel free to enable it if you would like. If you do, and
; you find problems, please do not report them.
; -------------------------------------
@ -22,30 +22,30 @@
;type=trunk ; This line is what marks this entry as a trunk.
;device=DAHDI/3 ; Map this trunk declaration to a specific device.
; NOTE: You can not just put any type of channel here.
; DAHDI channels can be directly used. IP trunks
; require some indirect configuration which is
; described in doc/asterisk.pdf.
; NOTE: You can not just put any type of channel here.
; DAHDI channels can be directly used. IP trunks
; require some indirect configuration which is
; described in doc/asterisk.pdf.
;autocontext=line1 ; This supports automatic generation of the dialplan entries
; if the autocontext option is used. Each trunk should have
; a unique context name. Then, in chan_dahdi.conf, this device
; should be configured to have incoming calls go to this context.
; if the autocontext option is used. Each trunk should have
; a unique context name. Then, in chan_dahdi.conf, this device
; should be configured to have incoming calls go to this context.
;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging
; it up as an unanswered call. The value is in seconds.
; it up as an unanswered call. The value is in seconds.
;barge=no ; If this option is set to "no", then no station will be
; allowed to join a call that is in progress. The default
; value is "yes".
; allowed to join a call that is in progress. The default
; value is "yes".
;hold=private ; This option configure hold permissions for this trunk.
; "open" - This means that any station can put this trunk
; on hold, and any station can retrieve it from
; hold. This is the default.
; "private" - This means that once a station puts the
; trunk on hold, no other station will be
; allowed to retrieve the call from hold.
; "open" - This means that any station can put this trunk
; on hold, and any station can retrieve it from
; hold. This is the default.
; "private" - This means that once a station puts the
; trunk on hold, no other station will be
; allowed to retrieve the call from hold.
;[line2]
;type=trunk
@ -60,9 +60,9 @@
;[line4]
;type=trunk
;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa
; application can be used to support IP trunks.
; See doc/asterisk.pdf on more information on how
; IP trunks work.
; application can be used to support IP trunks.
; See doc/asterisk.pdf on more information on how
; IP trunks work.
;autocontext=line4
; --------------------------------------
@ -76,48 +76,48 @@
;device=SIP/station1 ; Each station must be mapped to a device.
;autocontext=sla_stations ; This supports automatic generation of the dialplan entries if
; the autocontext option is used. All stations can use the same
; context without conflict. The device for this station should
; have its context configured to the same one listed here.
; the autocontext option is used. All stations can use the same
; context without conflict. The device for this station should
; have its context configured to the same one listed here.
;ringtimeout=10 ; Set a timeout for how long to allow the station to ring for an
; incoming call, in seconds.
; incoming call, in seconds.
;ringdelay=10 ; Set a time for how long to wait before beginning to ring this station
; once there is an incoming call, in seconds.
; once there is an incoming call, in seconds.
;hold=private ; This option configure hold permissions for this station. Note
; that if private hold is set in the trunk entry, that will override
; anything here. However, if a trunk has open hold access, but this
; station is set to private hold, then the private hold will be in
; effect.
; "open" - This means that once this station puts a call
; on hold, any other station is allowed to retrieve
; it. This is the default.
; "private" - This means that once this station puts a
; call on hold, no other station will be
; allowed to retrieve the call from hold.
; that if private hold is set in the trunk entry, that will override
; anything here. However, if a trunk has open hold access, but this
; station is set to private hold, then the private hold will be in
; effect.
; "open" - This means that once this station puts a call
; on hold, any other station is allowed to retrieve
; it. This is the default.
; "private" - This means that once this station puts a
; call on hold, no other station will be
; allowed to retrieve the call from hold.
;trunk=line1 ; Individually list all of the trunks that will appear on this station. This
; order is significant. It should be the same order as they appear on the
; phone. The order here defines the order of preference that the trunks will
; be used.
; order is significant. It should be the same order as they appear on the
; phone. The order here defines the order of preference that the trunks will
; be used.
;trunk=line2
;trunk=line3,ringdelay=5 ; A ring delay for the station can also be specified for a specific trunk.
; If a ring delay is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
; If a ring delay is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
;trunk=line4,ringtimeout=5 ; A ring timeout for the station can also be specified for a specific trunk.
; If a ring timeout is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
; If a ring timeout is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
;[station](!) ; When there are a lot of stations that are configured the same way,
; it is convenient to use a configuration template like this so that
; the common settings stay in one place.
; it is convenient to use a configuration template like this so that
; the common settings stay in one place.
;type=station
;autocontext=sla_stations
;trunk=line1

70
configs/telcordia-1.adsi
View File

@ -28,15 +28,15 @@ STATE "inactive" ; No active call
; Begin soft key definitions
;
KEY "CB_OH" IS "Block" OR "Call Block"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
SUBSCRIPT "offHook"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
SUBSCRIPT "offHook"
ENDKEY
KEY "CB" IS "Block" OR "Call Block"
SENDDTMF "*60"
SENDDTMF "*60"
ENDKEY
;
@ -44,38 +44,38 @@ ENDKEY
;
SUB "main" IS
IFEVENT NEARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
IFEVENT OFFHOOK THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "CB"
GOTO "offHook"
ENDIF
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "CB_OH"
ENDIF
IFEVENT CALLERID THEN
CLEAR
SHOWDISPLAY "newcall" AT 1
ENDIF
IFEVENT NEARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
IFEVENT OFFHOOK THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "CB"
GOTO "offHook"
ENDIF
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "CB_OH"
ENDIF
IFEVENT CALLERID THEN
CLEAR
SHOWDISPLAY "newcall" AT 1
ENDIF
ENDSUB
SUB "offHook" IS
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "ringing" AT 1
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "ringing" AT 1
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
ENDSUB
SUB "stableCall" IS

30
configs/unistim.conf.sample
View File

@ -14,29 +14,29 @@ port=5000 ; UDP port
;keepalive=120 ; in seconds, default = 120
;public_ip= ; if asterisk is behind a nat, specify your public IP
;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important
; informations. no (default), yes, tn.
; informations. no (default), yes, tn.
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -63,9 +63,9 @@ port=5000 ; UDP port
;mailbox=1234 ; Specify the mailbox number. Used by Message Waiting Indication
;linelabel="Support" ; Softkey label for the next line=> entry, 9 char max.
;extension=none ; Add an extension into the dialplan. Only valid in context specified previously.
; none=don't add (default), ask=prompt user, line=use the line number
; none=don't add (default), ask=prompt user, line=use the line number
;line => 100 ; Only one line by device is currently supported.
; Beware ! only bookmark and softkey entries are allowed after line=>
; Beware ! only bookmark and softkey entries are allowed after line=>
;bookmark=Hans C.@123 ; Use a softkey to dial 123. Name : 9 char max
;bookmark=Mailbox@011@54 ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63)
;bookmark=Test@*@USTM/violet ; Display an icon if violet is connected (dynamic), only for unistim device

22
configs/usbradio.conf.sample
View File

@ -30,23 +30,23 @@
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; USBRADIO channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The USBRADIO channel can't accept jitter,
; thus an enabled jitterbuffer on the receive USBRADIO side will always
; be used if the sending side can create jitter.
; USBRADIO channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The USBRADIO channel can't accept jitter,
; thus an enabled jitterbuffer on the receive USBRADIO side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; 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
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an USBRADIO
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

98
configs/voicemail.conf.sample
View File

@ -222,84 +222,84 @@ emaildateformat=%A, %B %d, %Y at %r
; tz=central ; Timezone from zonemessages below. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email
; attachfmt=wav49 ; Which format to attach to the email. Normally this is the
; first format specified in the format parameter above, but this
; option lets you customize the format sent to particular mailboxes.
; Useful if Windows users want wav49, but Linux users want gsm.
; [per-mailbox only]
; first format specified in the format parameter above, but this
; option lets you customize the format sent to particular mailboxes.
; Useful if Windows users want wav49, but Linux users want gsm.
; [per-mailbox only]
; saycid=yes ; Say the caller id information before the message. If not described,
; or set to no, it will be in the envelope
; or set to no, it will be in the envelope
; cidinternalcontexts=intern ; Internal Context for Name Playback instead of
; extension digits when saying caller id.
; extension digits when saying caller id.
; sayduration=no ; Turn on/off the duration information before the message. [ON by default]
; saydurationm=2 ; Specify the minimum duration to say. Default is 2 minutes
; dialout=fromvm ; Context to dial out from [option 4 from mailbox's advanced menu].
; If not specified, option 4 will not be listed and dialing out
; from within VoiceMailMain() will not be permitted.
; If not specified, option 4 will not be listed and dialing out
; from within VoiceMailMain() will not be permitted.
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
; VoiceMailMain() [option 5 from mailbox's advanced menu].
; If set to 'no', option 5 will not be listed.
; VoiceMailMain() [option 5 from mailbox's advanced menu].
; If set to 'no', option 5 will not be listed.
; searchcontexts=yes ; Current default behavior is to search only the default context
; if one is not specified. The older behavior was to search all contexts.
; This option restores the old behavior [DEFAULT=no]
; Note: If you have this option enabled, then you will be required to have
; unique mailbox names across all contexts. Otherwise, an ambiguity is created
; since it is impossible to know which mailbox to retrieve when one is requested.
; if one is not specified. The older behavior was to search all contexts.
; This option restores the old behavior [DEFAULT=no]
; Note: If you have this option enabled, then you will be required to have
; unique mailbox names across all contexts. Otherwise, an ambiguity is created
; since it is impossible to know which mailbox to retrieve when one is requested.
; callback=fromvm ; Context to call back from
; if not listed, calling the sender back will not be permitted
; if not listed, calling the sender back will not be permitted
; exitcontext=fromvm ; Context to go to on user exit such as * or 0
; The default is the current context.
; The default is the current context.
; review=yes ; Allow sender to review/rerecord their message before saving it [OFF by default
; operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to
; reach an operator. This option REQUIRES an 'o' extension in the
; same context (or in exitcontext, if set), as that is where the
; 0 key will send you. [OFF by default]
; reach an operator. This option REQUIRES an 'o' extension in the
; same context (or in exitcontext, if set), as that is where the
; 0 key will send you. [OFF by default]
; envelope=no ; Turn on/off envelope playback before message playback. [ON by default]
; This does NOT affect option 3,3 from the advanced options menu
; This does NOT affect option 3,3 from the advanced options menu
; delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only]
; This is intended for use with users who wish to receive their
; voicemail ONLY by email. Note: "deletevoicemail" is provided as an
; equivalent option for Realtime configuration.
; This is intended for use with users who wish to receive their
; voicemail ONLY by email. Note: "deletevoicemail" is provided as an
; equivalent option for Realtime configuration.
; volgain=0.0 ; Emails bearing the voicemail may arrive in a volume too
; quiet to be heard. This parameter allows you to specify how
; much gain to add to the message when sending a voicemail.
; NOTE: sox must be installed for this option to work.
; quiet to be heard. This parameter allows you to specify how
; much gain to add to the message when sending a voicemail.
; NOTE: sox must be installed for this option to work.
; nextaftercmd=yes ; Skips to the next message after hitting 7 or 9 to delete/save current message.
; [global option only at this time]
; [global option only at this time]
; forcename=yes ; Forces a new user to record their name. A new user is
; determined by the password being the same as
; the mailbox number. The default is "no".
; determined by the password being the same as
; the mailbox number. The default is "no".
; forcegreetings=no ; This is the same as forcename, except for recording
; greetings. The default is "no".
; greetings. The default is "no".
; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory
; The default is "no".
; The default is "no".
; tempgreetwarn=yes ; Remind the user that their temporary greeting is set
;messagewrap=no ; Enable next/last message to wrap around to
; first (from last) and last (from first) message
; The default is "no".
; first (from last) and last (from first) message
; The default is "no".
; minpassword=0 ; Enforce minimum password length
; vm-password=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "password"
; Customize which sound file is used instead of the default
; prompt that says: "password"
; vm-newpassword=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "Please enter your new password followed by
; the pound key."
; Customize which sound file is used instead of the default
; prompt that says: "Please enter your new password followed by
; the pound key."
; vm-passchanged=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "Your password has been changed."
; Customize which sound file is used instead of the default
; prompt that says: "Your password has been changed."
; vm-reenterpassword=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "Please re-enter your password followed by
; the pound key"
; Customize which sound file is used instead of the default
; prompt that says: "Please re-enter your password followed by
; the pound key"
; vm-mismatch=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "The passwords you entered and re-entered
; did not match. Please try again."
; Customize which sound file is used instead of the default
; prompt that says: "The passwords you entered and re-entered
; did not match. Please try again."
; vm-invalid-password=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: ...
; Customize which sound file is used instead of the default
; prompt that says: ...
; listen-control-forward-key=# ; Customize the key that fast-forwards message playback
; listen-control-reverse-key=* ; Customize the key that rewinds message playback
; listen-control-pause-key=0 ; Customize the key that pauses/unpauses message playback