diff options
| author | seanbright <seanbright@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-05-28 14:32:03 +0000 |
|---|---|---|
| committer | seanbright <seanbright@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-05-28 14:32:03 +0000 |
| commit | 7f7cfd42e9d0e77e3d1ea18732595c1794bf46ea (patch) | |
| tree | f290e7e7ae143382811fc8ff32a52618ae0801bf | |
| parent | 0a45784168c75c3f5387fd7df1fef307e4797195 (diff) | |
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197528 f38db490-d61c-443f-a65b-d21fe96a405b
39 files changed, 1237 insertions, 1237 deletions
diff --git a/configs/alarmreceiver.conf.sample b/configs/alarmreceiver.conf.sample index bf767dea3..0ad23f8fc 100644 --- a/configs/alarmreceiver.conf.sample +++ b/configs/alarmreceiver.conf.sample @@ -10,7 +10,7 @@ ; ; Specify a timestamp format for the metadata section of the event files ; Default is %a %b %d, %Y @ %H:%M:%S %Z - + timestampformat = %a %b %d, %Y @ %H:%M:%S %Z ; diff --git a/configs/alsa.conf.sample b/configs/alsa.conf.sample index f55030618..33c5a3fa8 100644 --- a/configs/alsa.conf.sample +++ b/configs/alsa.conf.sample @@ -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". ;----------------------------------------------------------------------------------- diff --git a/configs/amd.conf.sample b/configs/amd.conf.sample index ce4808a0c..e25c18e18 100644 --- a/configs/amd.conf.sample +++ b/configs/amd.conf.sample @@ -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 diff --git a/configs/asterisk.adsi b/configs/asterisk.adsi index a275502ac..396de2c75 100644 --- a/configs/asterisk.adsi +++ b/configs/asterisk.adsi @@ -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 diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample index a15762c43..76771fb3e 100644 --- a/configs/chan_dahdi.conf.sample +++ b/configs/chan_dahdi.conf.sample @@ -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". ;----------------------------------------------------------------------------------- diff --git a/configs/cli_aliases.conf.sample b/configs/cli_aliases.conf.sample index 7743a47d6..cc1e2e6d3 100644 --- a/configs/cli_aliases.conf.sample +++ b/configs/cli_aliases.conf.sample @@ -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, diff --git a/configs/cli_permissions.conf.sample b/configs/cli_permissions.conf.sample index 4a6973f50..7cbad88f3 100644 --- a/configs/cli_permissions.conf.sample +++ b/configs/cli_permissions.conf.sample @@ -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. ; diff --git a/configs/console.conf.sample b/configs/console.conf.sample index ff58605a3..d7e586a6b 100644 --- a/configs/console.conf.sample +++ b/configs/console.conf.sample @@ -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. diff --git a/configs/dnsmgr.conf.sample b/configs/dnsmgr.conf.sample index e34dbcf0a..a2939dc10 100644 --- a/configs/dnsmgr.conf.sample +++ b/configs/dnsmgr.conf.sample @@ -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)
\ No newline at end of file +; default is 300 (5 minutes)
\ No newline at end of file diff --git a/configs/extensions.ael.sample b/configs/extensions.ael.sample index 21680a4db..c7720290a 100644 --- a/configs/extensions.ael.sample +++ b/configs/extensions.ael.sample @@ -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 @@ globals { // // 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 - // - - switches { - DUNDi/e164; - }; +// +// Just a wrapper for the switch +// + +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 ${exten}|1; +return; }; // @@ -186,7 +186,7 @@ macro ael-dundi-e164(exten) { // up, please go to www.gnophone.com or www.iaxtel.com // context ael-iaxtel700 { - _91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel); +_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel); }; // @@ -196,91 +196,91 @@ context ael-iaxtel700 { // to be on-line or else dialing can be severly delayed. // context ael-iaxprovider { - switches { - // IAX2/user:[key]@myserver/mycontext; - }; +switches { +// IAX2/user:[key]@myserver/mycontext; +}; }; context ael-trunkint { - // - // International long distance through trunk - // - includes { - ael-dundi-e164-lookup; - }; - _9011. => { - &ael-dundi-e164(${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:${TRUNKMSD}}); +}; }; context ael-trunktollfree { - // - // Long distance context accessed through trunk interface - // - - _91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); - _91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); - _91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); - _91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); +// +// Long distance context accessed through trunk interface +// + +_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); +_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); +_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); +_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); }; context ael-international { - // - // Master context for international long distance - // - ignorepat => 9; - includes { - ael-longdistance; - ael-trunkint; - }; +// +// Master context for international long distance +// +ignorepat => 9; +includes { +ael-longdistance; +ael-trunkint; +}; }; context ael-longdistance { - // - // Master context for long distance - // - ignorepat => 9; - includes { - ael-local; - ael-trunkld; - }; +// +// Master context for long distance +// +ignorepat => 9; +includes { +ael-local; +ael-trunkld; +}; }; context ael-local { - // - // Master context for local, toll-free, and iaxtel calls only - // - ignorepat => 9; - includes { - ael-default; - ael-trunklocal; - ael-iaxtel700; - ael-trunktollfree; - ael-iaxprovider; - }; +// +// Master context for local, toll-free, and iaxtel calls only +// +ignorepat => 9; +includes { +ael-default; +ael-trunklocal; +ael-iaxtel700; +ael-trunktollfree; +ael-iaxprovider; +}; }; // @@ -306,69 +306,69 @@ context ael-local { macro ael-std-exten-ael( ext , dev ) { - Dial(${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(${EXTEN}, "IAX2"); +8500 => { +VoicemailMain(); +goto s|instructions; +}; +# => { +Playback(demo-thanks); +Hangup(); +}; +t => goto #|1; +i => Playback(invalid); }; @@ -383,9 +383,9 @@ context ael-default { // By default we include the demo. In a production system, you // probably don't want to have the demo there. - includes { - ael-demo; - }; +includes { +ael-demo; +}; // // Extensions like the two below can be used for FWD, Nikotel, sipgate etc. // Note that you must have a [sipprovider] section in sip.conf whereas diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample index 9e6207d22..230576d45 100644 --- a/configs/extensions.conf.sample +++ b/configs/extensions.conf.sample @@ -430,7 +430,7 @@ exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER) exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start exten => stdexten-BUSY,1,Voicemail(${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) diff --git a/configs/extensions.lua.sample b/configs/extensions.lua.sample index 44b9b81b5..0bbb3aef1 100644 --- a/configs/extensions.lua.sample +++ b/configs/extensions.lua.sample @@ -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; - - ["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; - - ["1234"] = function() - app.playback("transfer", "skip") - -- do a dial here - end; - - ["1235"] = function() - app.voicemail("1234", "u") - 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; - - ["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; - - ["8500"] = function() - app.voicemailmain() - demo_instruct() - end; - - }; - - default = { - -- by default, do the demo - include = {"demo"}; - }; - - ["local"] = { - ["_NXXXXXX"] = outgoing_local; - }; +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; + +["1000"] = function() +app.goto("default", "s", 1) +end; + +["1234"] = function() +app.playback("transfer", "skip") +-- do a dial here +end; + +["1235"] = function() +app.voicemail("1234", "u") +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; + +["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; + +["8500"] = function() +app.voicemailmain() +demo_instruct() +end; + +}; + +default = { +-- by default, do the demo +include = {"demo"}; +}; + +["local"] = { +["_NXXXXXX"] = outgoing_local; +}; } diff --git a/configs/features.conf.sample b/configs/features.conf.sample index abd0d5d2d..83aa69643 100644 --- a/configs/features.conf.sample +++ b/configs/features.conf.sample @@ -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 diff --git a/configs/func_odbc.conf.sample b/configs/func_odbc.conf.sample index 1bc11be2e..2b67e5396 100644 --- a/configs/func_odbc.conf.sample +++ b/configs/func_odbc.conf.sample @@ -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 diff --git a/configs/gtalk.conf.sample b/configs/gtalk.conf.sample index ad089b208..f3dd3f830 100644 --- a/configs/gtalk.conf.sample +++ b/configs/gtalk.conf.sample @@ -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. ; diff --git a/configs/h323.conf.sample b/configs/h323.conf.sample index 5be321f33..c2e5db328 100644 --- a/configs/h323.conf.sample +++ b/configs/h323.conf.sample @@ -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". ;----------------------------------------------------------------------------------- diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample index df7796f2c..259fe626b 100644 --- a/configs/iax.conf.sample +++ b/configs/iax.conf.sample @@ -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. diff --git a/configs/jabber.conf.sample b/configs/jabber.conf.sample index 6cfb755bd..2990d8e91 100644 --- a/configs/jabber.conf.sample +++ b/configs/jabber.conf.sample @@ -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. diff --git a/configs/jingle.conf.sample b/configs/jingle.conf.sample index ad089b208..f3dd3f830 100644 --- a/configs/jingle.conf.sample +++ b/configs/jingle.conf.sample @@ -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. ; diff --git a/configs/manager.conf.sample b/configs/manager.conf.sample index 425ce4ca2..855b9e6bc 100644 --- a/configs/manager.conf.sample +++ b/configs/manager.conf.sample @@ -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 diff --git a/configs/meetme.conf.sample b/configs/meetme.conf.sample index 3eb3a82a5..05bcb893f 100644 --- a/configs/meetme.conf.sample +++ b/configs/meetme.conf.sample @@ -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 diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample index 104891e8a..01c8fe77c 100644 --- a/configs/mgcp.conf.sample +++ b/configs/mgcp.conf.sample @@ -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 diff --git a/configs/minivm.conf.sample b/configs/minivm.conf.sample index 21d18e0c6..0e29dd96d 100644 --- a/configs/minivm.conf.sample +++ b/configs/minivm.conf.sample @@ -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] diff --git a/configs/misdn.conf.sample b/configs/misdn.conf.sample index f4ca486e9..08fb288f3 100644 --- a/configs/misdn.conf.sample +++ b/configs/misdn.conf.sample @@ -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". ;----------------------------------------------------------------------------------- diff --git a/configs/musiconhold.conf.sample b/configs/musiconhold.conf.sample index 8ccc851e4..39df862bf 100644 --- a/configs/musiconhold.conf.sample +++ b/configs/musiconhold.conf.sample @@ -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: diff --git a/configs/oss.conf.sample b/configs/oss.conf.sample index 48f0ced90..f0ed94ea6 100644 --- a/configs/oss.conf.sample +++ b/configs/oss.conf.sample @@ -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 - - ; 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 = - - ; 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> - - ; 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. - - ; 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. - - ; 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. - - ; 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". - ;----------------------------------------------------------------------------------- +; 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 + +; 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 + +; 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 + +; 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 + +;------------------------------ 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. + +; 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. + +; 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, ... - - ; 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 - - bitrate = 60000 ; rate told to ffmpeg. - fps = 5 ; frames per second from the source. - ; qmin = 3 ; quantizer value passed to the encoder. +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, ... + +; 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. ; 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 @@ ; 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 diff --git a/configs/phoneprov.conf.sample b/configs/phoneprov.conf.sample index cb1acec42..f3df08c49 100644 --- a/configs/phoneprov.conf.sample +++ b/configs/phoneprov.conf.sample @@ -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 diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample index cefc4abaf..fb45a2e9e 100644 --- a/configs/queues.conf.sample +++ b/configs/queues.conf.sample @@ -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 diff --git a/configs/res_odbc.conf.sample b/configs/res_odbc.conf.sample index 217cd2ffc..85bd8f45a 100644 --- a/configs/res_odbc.conf.sample +++ b/configs/res_odbc.conf.sample @@ -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. diff --git a/configs/rpt.conf.sample b/configs/rpt.conf.sample index 823672438..871793d65 100644 --- a/configs/rpt.conf.sample +++ b/configs/rpt.conf.sample @@ -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] diff --git a/configs/rtp.conf.sample b/configs/rtp.conf.sample index f90ed890d..615b6fe46 100644 --- a/configs/rtp.conf.sample +++ b/configs/rtp.conf.sample @@ -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 diff --git a/configs/say.conf.sample b/configs/say.conf.sample index e8b8ed7ae..f592b780a 100644 --- a/configs/say.conf.sample +++ b/configs/say.conf.sample @@ -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: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: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:${SAY:1} +_[n]um:0. => num:${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:${SAY:3} + +_[n]um:XXXXXXX => num:${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:${SAY:3} + +; enumeration +_e[n]um:X => digits/h-${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:${SAY:1} - - _[n]um:1XXX => digits/thousand, num:${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} - - _datetime::. => date:AdBY 'digits/at' IMp:${SAY} - _date::. => date:AdBY:${SAY} - _time::. => date:IMp:${SAY} +_[n]um:0. => num:${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:${SAY:3} + +_[n]um:1XXXXXX => num:${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} [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:${SAY:3} - - _datetime::. => date:AdBY 'digits/at' IMp:${SAY} - _date::. => date:AdBY:${SAY} - _time::. => date:IMp:${SAY} +_[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} [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:${SAY:1} - - _[n]um:XXXX => num:${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:${SAY:3} - - _[n]um:XXXXXXXXXX => num:${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} +_[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:${SAY:3} + +_[n]um:XXXXXXX => num:${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:${SAY:3} + +_datetime::. => date:YBdA k 'ora' M 'perc':${SAY} +_date::. => date:YBdA:${SAY} +_time::. => date:k 'ora' M 'perc':${SAY} diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index f9e656419..862b482d4 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -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 ----------------------- ; @@ -533,12 +533,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; Note that in this example, the optional authuser and secret portions have ; been left blank because we have specified a port in the user section - + ;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. - - ; 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). +; 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). ;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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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. diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample index a7b188c45..f5613ac21 100644 --- a/configs/skinny.conf.sample +++ b/configs/skinny.conf.sample @@ -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] diff --git a/configs/sla.conf.sample b/configs/sla.conf.sample index fc8865424..9fdb3f336 100644 --- a/configs/sla.conf.sample +++ b/configs/sla.conf.sample @@ -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 diff --git a/configs/telcordia-1.adsi b/configs/telcordia-1.adsi index 1486aa95e..96eb1db21 100644 --- a/configs/telcordia-1.adsi +++ b/configs/telcordia-1.adsi @@ -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 diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample index 649737317..2b61a8646 100644 --- a/configs/unistim.conf.sample +++ b/configs/unistim.conf.sample @@ -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 diff --git a/configs/usbradio.conf.sample b/configs/usbradio.conf.sample index 5ba9815ca..2b62ea809 100644 --- a/configs/usbradio.conf.sample +++ b/configs/usbradio.conf.sample @@ -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". ;----------------------------------------------------------------------------------- diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample index 3606b14b8..5d6397608 100644 --- a/configs/voicemail.conf.sample +++ b/configs/voicemail.conf.sample @@ -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 |