Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
parent
7f7cfd42e9
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a22b4735e5
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@ -39,23 +39,23 @@ extension=s
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; ALSA channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The ALSA channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive ALSA side will always
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; be used if the sending side can create jitter.
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; ALSA channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The ALSA channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive ALSA side will always
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; be used if the sending side can create jitter.
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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@ -4,15 +4,15 @@
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[general]
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initial_silence = 2500 ; Maximum silence duration before the greeting.
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; If exceeded then MACHINE.
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; If exceeded then MACHINE.
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greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
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after_greeting_silence = 800 ; Silence after detecting a greeting.
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; If exceeded then HUMAN
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; If exceeded then HUMAN
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total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
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; on a HUMAN or MACHINE
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; on a HUMAN or MACHINE
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min_word_length = 100 ; Minimum duration of Voice to considered as a word
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between_words_silence = 50 ; Minimum duration of silence after a word to consider
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; the audio what follows as a new word
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; the audio what follows as a new word
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maximum_number_of_words = 3 ; Maximum number of words in the greeting.
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; If exceeded then MACHINE
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; If exceeded then MACHINE
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silence_threshold = 256
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@ -35,39 +35,39 @@ DISPLAY "empty" IS "asdf"
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; Begin soft key definitions
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;
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KEY "callfwd" IS "CallFwd" OR "Call Forward"
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OFFHOOK
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VOICEMODE
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WAITDIALTONE
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SENDDTMF "*60"
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GOTO "offHook"
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OFFHOOK
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VOICEMODE
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WAITDIALTONE
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SENDDTMF "*60"
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GOTO "offHook"
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ENDKEY
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KEY "vmail_OH" IS "VMail" OR "Voicemail"
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OFFHOOK
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VOICEMODE
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WAITDIALTONE
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SENDDTMF "8500"
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OFFHOOK
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VOICEMODE
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WAITDIALTONE
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SENDDTMF "8500"
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ENDKEY
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KEY "vmail" IS "VMail" OR "Voicemail"
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SENDDTMF "8500"
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SENDDTMF "8500"
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ENDKEY
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KEY "backspace" IS "BackSpc" OR "Backspace"
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BACKSPACE
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BACKSPACE
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ENDKEY
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KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
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SENDDTMF "*70"
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SETFLAG "nocallwaiting"
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SHOWDISPLAY "cwdisabled" AT 4
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TIMERCLEAR
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TIMERSTART 1
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SENDDTMF "*70"
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SETFLAG "nocallwaiting"
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SHOWDISPLAY "cwdisabled" AT 4
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TIMERCLEAR
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TIMERSTART 1
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ENDKEY
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KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
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SENDDTMF "*67"
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SETFLAG "nocallwaiting"
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SENDDTMF "*67"
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SETFLAG "nocallwaiting"
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ENDKEY
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;
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@ -75,85 +75,85 @@ ENDKEY
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;
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SUB "main" IS
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IFEVENT NEARANSWER THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "talkingto" AT 2 NOUPDATE
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SHOWDISPLAY "callname" AT 3
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SHOWDISPLAY "callnum" AT 4
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GOTO "stableCall"
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ENDIF
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IFEVENT OFFHOOK THEN
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CLEAR
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CLEARFLAG "nocallwaiting"
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CLEARDISPLAY
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SHOWDISPLAY "titles" AT 1
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SHOWKEYS "vmail"
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SHOWKEYS "cidblock"
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SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
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GOTO "offHook"
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ENDIF
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IFEVENT IDLE THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1
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SHOWKEYS "vmail_OH"
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ENDIF
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IFEVENT CALLERID THEN
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CLEAR
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IFEVENT NEARANSWER THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "talkingto" AT 2 NOUPDATE
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SHOWDISPLAY "callname" AT 3
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SHOWDISPLAY "callnum" AT 4
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GOTO "stableCall"
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ENDIF
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IFEVENT OFFHOOK THEN
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CLEAR
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CLEARFLAG "nocallwaiting"
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CLEARDISPLAY
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SHOWDISPLAY "titles" AT 1
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SHOWKEYS "vmail"
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SHOWKEYS "cidblock"
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SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
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GOTO "offHook"
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ENDIF
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IFEVENT IDLE THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1
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SHOWKEYS "vmail_OH"
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ENDIF
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IFEVENT CALLERID THEN
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CLEAR
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; SHOWDISPLAY "titles" AT 1 NOUPDATE
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; SHOWDISPLAY "incoming" AT 2 NOUPDATE
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SHOWDISPLAY "callname" AT 3 NOUPDATE
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SHOWDISPLAY "callnum" AT 4
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ENDIF
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IFEVENT RING THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "incoming" AT 2
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ENDIF
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IFEVENT ENDOFRING THEN
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SHOWDISPLAY "missedcall" AT 2
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CLEAR
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SHOWDISPLAY "titles" AT 1
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SHOWKEYS "vmail_OH"
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ENDIF
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IFEVENT TIMER THEN
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CLEAR
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SHOWDISPLAY "empty" AT 4
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ENDIF
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SHOWDISPLAY "callname" AT 3 NOUPDATE
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SHOWDISPLAY "callnum" AT 4
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ENDIF
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IFEVENT RING THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "incoming" AT 2
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ENDIF
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IFEVENT ENDOFRING THEN
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SHOWDISPLAY "missedcall" AT 2
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CLEAR
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SHOWDISPLAY "titles" AT 1
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SHOWKEYS "vmail_OH"
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ENDIF
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IFEVENT TIMER THEN
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CLEAR
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SHOWDISPLAY "empty" AT 4
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ENDIF
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ENDSUB
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SUB "offHook" IS
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IFEVENT FARRING THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "ringing" AT 2 NOUPDATE
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SHOWDISPLAY "callname" at 3 NOUPDATE
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SHOWDISPLAY "callnum" at 4
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ENDIF
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IFEVENT FARANSWER THEN
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CLEAR
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SHOWDISPLAY "talkingto" AT 2
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GOTO "stableCall"
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ENDIF
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IFEVENT BUSY THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "busy" AT 2 NOUPDATE
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SHOWDISPLAY "callname" at 3 NOUPDATE
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SHOWDISPLAY "callnum" at 4
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ENDIF
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IFEVENT REORDER THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "reorder" AT 2 NOUPDATE
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SHOWDISPLAY "callname" at 3 NOUPDATE
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SHOWDISPLAY "callnum" at 4
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ENDIF
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IFEVENT FARRING THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "ringing" AT 2 NOUPDATE
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SHOWDISPLAY "callname" at 3 NOUPDATE
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SHOWDISPLAY "callnum" at 4
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ENDIF
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IFEVENT FARANSWER THEN
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CLEAR
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SHOWDISPLAY "talkingto" AT 2
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GOTO "stableCall"
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ENDIF
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IFEVENT BUSY THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "busy" AT 2 NOUPDATE
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SHOWDISPLAY "callname" at 3 NOUPDATE
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SHOWDISPLAY "callnum" at 4
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ENDIF
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IFEVENT REORDER THEN
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CLEAR
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SHOWDISPLAY "titles" AT 1 NOUPDATE
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SHOWDISPLAY "reorder" AT 2 NOUPDATE
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SHOWDISPLAY "callname" at 3 NOUPDATE
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SHOWDISPLAY "callnum" at 4
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ENDIF
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ENDSUB
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SUB "stableCall" IS
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IFEVENT REORDER THEN
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SHOWDISPLAY "callended" AT 2
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ENDIF
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IFEVENT REORDER THEN
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SHOWDISPLAY "callended" AT 2
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ENDIF
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ENDSUB
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@ -581,9 +581,9 @@ pickupgroup=1
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; Channel variable to be set for all calls from this channel
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;setvar=CHANNEL=42
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;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
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; cause the given audio file to
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; be played upon completion of
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; an attended transfer.
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; cause the given audio file to
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; be played upon completion of
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; an attended transfer.
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;
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; Specify whether the channel should be answered immediately or if the simple
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@ -792,23 +792,23 @@ pickupgroup=1
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;
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The DAHDI channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive DAHDI side will always
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; be used if the sending side can create jitter.
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; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The DAHDI channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive DAHDI side will always
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; be used if the sending side can create jitter.
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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@ -13,8 +13,8 @@ template = friendly ; By default, include friendly aliases
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;template = asterisk12 ; Asterisk 1.2 style syntax
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;template = asterisk14 ; Asterisk 1.4 style syntax
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;template = individual_custom ; see [individual_custom] example below which
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; includes a list of aliases from an external
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; file
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; includes a list of aliases from an external
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; file
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; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
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@ -23,7 +23,7 @@
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[general]
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default_perm=permit ; To leave asterisk working as normal
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; we should set this parameter to 'permit'
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; we should set this parameter to 'permit'
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;
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; Follows the per-users permissions configs.
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;
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@ -34,7 +34,7 @@
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; The default is "no".
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;
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;overridecontext = no ; if 'no', the last @ will start the context
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; if 'yes' the whole string is an extension.
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; if 'yes' the whole string is an extension.
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; Default Music on Hold class to use when this channel is placed on hold in
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@ -46,23 +46,23 @@
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; Console channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The Console channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive Console side will always
|
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; be used if the sending side can create jitter.
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; Console channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
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; side can not accept jitter. The Console channel can't accept jitter,
|
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; thus an enabled jitterbuffer on the receive Console side will always
|
||||
; be used if the sending side can create jitter.
|
||||
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||||
; 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
|
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; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
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||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a 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.
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||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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||||
;-----------------------------------------------------------------------------------
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||||
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@ -76,8 +76,8 @@
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[default]
|
||||
input_device = default ; When configuring an input device and output device,
|
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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.
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||||
; list available" CLI command. If you say "default", the
|
||||
; system default input and output devices will be used.
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autoanswer = no
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context = default
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extension = s
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@ -86,5 +86,5 @@ language = en
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overridecontext = no
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mohinterpret = default
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active = yes ; This option should only be set for one console.
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||||
; It means that it is the active console to be
|
||||
; used from the Asterisk CLI.
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||||
; It means that it is the active console to be
|
||||
; used from the Asterisk CLI.
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||||
|
|
|
@ -1,5 +1,5 @@
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|||
[general]
|
||||
;enable=yes ; enable creation of managed DNS lookups
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||||
; default is 'no'
|
||||
; default is 'no'
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||||
;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds
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||||
; default is 300 (5 minutes)
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||||
; default is 300 (5 minutes)
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|
@ -19,28 +19,28 @@
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|||
//
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||||
|
||||
globals {
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||||
CONSOLE="Console/dsp"; // Console interface for demo
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||||
//CONSOLE=DAHDI/1
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||||
//CONSOLE=Phone/phone0
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||||
IAXINFO=guest; // IAXtel username/password
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||||
//IAXINFO="myuser:mypass";
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||||
TRUNK="DAHDI/G2"; // Trunk interface
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||||
//
|
||||
// Note the 'G2' in the TRUNK variable above. It specifies which group (defined
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||||
// in dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
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||||
// the specified group. The four possible options are:
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||||
//
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||||
// g: select the lowest-numbered non-busy DAHDI channel
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||||
// (aka. ascending sequential hunt group).
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||||
// G: select the highest-numbered non-busy DAHDI channel
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||||
// (aka. descending sequential hunt group).
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||||
// r: use a round-robin search, starting at the next highest channel than last
|
||||
// time (aka. ascending rotary hunt group).
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||||
// R: use a round-robin search, starting at the next lowest channel than last
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||||
// time (aka. descending rotary hunt group).
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||||
//
|
||||
TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
|
||||
//TRUNK=IAX2/user:pass@provider
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||||
CONSOLE="Console/dsp"; // Console interface for demo
|
||||
//CONSOLE=DAHDI/1
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||||
//CONSOLE=Phone/phone0
|
||||
IAXINFO=guest; // IAXtel username/password
|
||||
//IAXINFO="myuser:mypass";
|
||||
TRUNK="DAHDI/G2"; // Trunk interface
|
||||
//
|
||||
// Note the 'G2' in the TRUNK variable above. It specifies which group (defined
|
||||
// in dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
|
||||
// the specified group. The four possible options are:
|
||||
//
|
||||
// g: select the lowest-numbered non-busy DAHDI channel
|
||||
// (aka. ascending sequential hunt group).
|
||||
// G: select the highest-numbered non-busy DAHDI channel
|
||||
// (aka. descending sequential hunt group).
|
||||
// r: use a round-robin search, starting at the next highest channel than last
|
||||
// time (aka. ascending rotary hunt group).
|
||||
// R: use a round-robin search, starting at the next lowest channel than last
|
||||
// time (aka. descending rotary hunt group).
|
||||
//
|
||||
TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
|
||||
//TRUNK=IAX2/user:pass@provider
|
||||
};
|
||||
|
||||
//
|
||||
|
@ -110,61 +110,61 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
|
|||
//
|
||||
//
|
||||
context ael-dundi-e164-canonical {
|
||||
//
|
||||
// List canonical entries here
|
||||
//
|
||||
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
|
||||
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
|
||||
//
|
||||
// List canonical entries here
|
||||
//
|
||||
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
|
||||
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
|
||||
};
|
||||
|
||||
context ael-dundi-e164-customers {
|
||||
//
|
||||
// If you are an ITSP or Reseller, list your customers here.
|
||||
//
|
||||
//_12564286000 => Dial(SIP/customer1);
|
||||
//_12564286001 => Dial(IAX2/customer2);
|
||||
//
|
||||
// If you are an ITSP or Reseller, list your customers here.
|
||||
//
|
||||
//_12564286000 => Dial(SIP/customer1);
|
||||
//_12564286001 => Dial(IAX2/customer2);
|
||||
};
|
||||
|
||||
context ael-dundi-e164-via-pstn {
|
||||
//
|
||||
// If you are freely delivering calls to the PSTN, list them here
|
||||
//
|
||||
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
|
||||
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
|
||||
//
|
||||
// If you are freely delivering calls to the PSTN, list them here
|
||||
//
|
||||
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
|
||||
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
|
||||
};
|
||||
|
||||
context ael-dundi-e164-local {
|
||||
//
|
||||
// Context to put your dundi IAX2 or SIP user in for
|
||||
// full access
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-canonical;
|
||||
ael-dundi-e164-customers;
|
||||
ael-dundi-e164-via-pstn;
|
||||
};
|
||||
//
|
||||
// Context to put your dundi IAX2 or SIP user in for
|
||||
// full access
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-canonical;
|
||||
ael-dundi-e164-customers;
|
||||
ael-dundi-e164-via-pstn;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-dundi-e164-switch {
|
||||
//
|
||||
// Just a wrapper for the switch
|
||||
//
|
||||
//
|
||||
// Just a wrapper for the switch
|
||||
//
|
||||
|
||||
switches {
|
||||
DUNDi/e164;
|
||||
};
|
||||
switches {
|
||||
DUNDi/e164;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-dundi-e164-lookup {
|
||||
//
|
||||
// Locally to lookup, try looking for a local E.164 solution
|
||||
// then try DUNDi if we don't have one.
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-local;
|
||||
ael-dundi-e164-switch;
|
||||
};
|
||||
//
|
||||
//
|
||||
// Locally to lookup, try looking for a local E.164 solution
|
||||
// then try DUNDi if we don't have one.
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-local;
|
||||
ael-dundi-e164-switch;
|
||||
};
|
||||
//
|
||||
};
|
||||
|
||||
//
|
||||
|
@ -175,8 +175,8 @@ macro ael-dundi-e164(exten) {
|
|||
//
|
||||
// ARG1 is the extension to Dial
|
||||
//
|
||||
goto ${exten}|1;
|
||||
return;
|
||||
goto ${exten}|1;
|
||||
return;
|
||||
};
|
||||
|
||||
//
|
||||
|
@ -186,7 +186,7 @@ return;
|
|||
// up, please go to www.gnophone.com or www.iaxtel.com
|
||||
//
|
||||
context ael-iaxtel700 {
|
||||
_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
|
||||
_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
|
||||
};
|
||||
|
||||
//
|
||||
|
@ -196,91 +196,91 @@ _91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
|
|||
// to be on-line or else dialing can be severly delayed.
|
||||
//
|
||||
context ael-iaxprovider {
|
||||
switches {
|
||||
// IAX2/user:[key]@myserver/mycontext;
|
||||
};
|
||||
switches {
|
||||
// IAX2/user:[key]@myserver/mycontext;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunkint {
|
||||
//
|
||||
// International long distance through trunk
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-lookup;
|
||||
};
|
||||
_9011. => {
|
||||
&ael-dundi-e164(${EXTEN:4});
|
||||
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
//
|
||||
// International long distance through trunk
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-lookup;
|
||||
};
|
||||
_9011. => {
|
||||
&ael-dundi-e164(${EXTEN:4});
|
||||
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunkld {
|
||||
//
|
||||
// Long distance context accessed through trunk
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-lookup;
|
||||
};
|
||||
_91NXXNXXXXXX => {
|
||||
&ael-dundi-e164(${EXTEN:1});
|
||||
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
//
|
||||
// Long distance context accessed through trunk
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-lookup;
|
||||
};
|
||||
_91NXXNXXXXXX => {
|
||||
&ael-dundi-e164(${EXTEN:1});
|
||||
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunklocal {
|
||||
//
|
||||
// Local seven-digit dialing accessed through trunk interface
|
||||
//
|
||||
_9NXXXXXX => {
|
||||
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
//
|
||||
// Local seven-digit dialing accessed through trunk interface
|
||||
//
|
||||
_9NXXXXXX => {
|
||||
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunktollfree {
|
||||
//
|
||||
// Long distance context accessed through trunk interface
|
||||
//
|
||||
//
|
||||
// Long distance context accessed through trunk interface
|
||||
//
|
||||
|
||||
_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
|
||||
};
|
||||
|
||||
context ael-international {
|
||||
//
|
||||
// Master context for international long distance
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-longdistance;
|
||||
ael-trunkint;
|
||||
};
|
||||
//
|
||||
// Master context for international long distance
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-longdistance;
|
||||
ael-trunkint;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-longdistance {
|
||||
//
|
||||
// Master context for long distance
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-local;
|
||||
ael-trunkld;
|
||||
};
|
||||
//
|
||||
// Master context for long distance
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-local;
|
||||
ael-trunkld;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-local {
|
||||
//
|
||||
// Master context for local, toll-free, and iaxtel calls only
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-default;
|
||||
ael-trunklocal;
|
||||
ael-iaxtel700;
|
||||
ael-trunktollfree;
|
||||
ael-iaxprovider;
|
||||
};
|
||||
//
|
||||
// Master context for local, toll-free, and iaxtel calls only
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-default;
|
||||
ael-trunklocal;
|
||||
ael-iaxtel700;
|
||||
ael-trunktollfree;
|
||||
ael-iaxprovider;
|
||||
};
|
||||
};
|
||||
|
||||
//
|
||||
|
@ -306,69 +306,69 @@ ael-iaxprovider;
|
|||
|
||||
|
||||
macro ael-std-exten-ael( ext , dev ) {
|
||||
Dial(${dev}/${ext},20);
|
||||
switch(${DIALSTATUS}) {
|
||||
case BUSY:
|
||||
Voicemail(${ext},b);
|
||||
break;
|
||||
default:
|
||||
Voicemail(${ext},u);
|
||||
};
|
||||
catch a {
|
||||
VoiceMailMain(${ext});
|
||||
return;
|
||||
};
|
||||
return;
|
||||
Dial(${dev}/${ext},20);
|
||||
switch(${DIALSTATUS}) {
|
||||
case BUSY:
|
||||
Voicemail(${ext},b);
|
||||
break;
|
||||
default:
|
||||
Voicemail(${ext},u);
|
||||
};
|
||||
catch a {
|
||||
VoiceMailMain(${ext});
|
||||
return;
|
||||
};
|
||||
return;
|
||||
};
|
||||
|
||||
context ael-demo {
|
||||
s => {
|
||||
Wait(1);
|
||||
Answer();
|
||||
Set(TIMEOUT(digit)=5);
|
||||
Set(TIMEOUT(response)=10);
|
||||
s => {
|
||||
Wait(1);
|
||||
Answer();
|
||||
Set(TIMEOUT(digit)=5);
|
||||
Set(TIMEOUT(response)=10);
|
||||
restart:
|
||||
Background(demo-congrats);
|
||||
Background(demo-congrats);
|
||||
instructions:
|
||||
for (x=0; ${x} < 3; x=${x} + 1) {
|
||||
Background(demo-instruct);
|
||||
WaitExten();
|
||||
};
|
||||
};
|
||||
2 => {
|
||||
Background(demo-moreinfo);
|
||||
goto s|instructions;
|
||||
};
|
||||
3 => {
|
||||
Set(LANGUAGE()=fr);
|
||||
goto s|restart;
|
||||
};
|
||||
1000 => {
|
||||
goto ael-default|s|1;
|
||||
};
|
||||
500 => {
|
||||
Playback(demo-abouttotry);
|
||||
Dial(IAX2/guest@misery.digium.com/s@default);
|
||||
Playback(demo-nogo);
|
||||
goto s|instructions;
|
||||
};
|
||||
600 => {
|
||||
Playback(demo-echotest);
|
||||
Echo();
|
||||
Playback(demo-echodone);
|
||||
goto s|instructions;
|
||||
};
|
||||
_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
|
||||
8500 => {
|
||||
VoicemailMain();
|
||||
goto s|instructions;
|
||||
};
|
||||
# => {
|
||||
Playback(demo-thanks);
|
||||
Hangup();
|
||||
};
|
||||
t => goto #|1;
|
||||
i => Playback(invalid);
|
||||
for (x=0; ${x} < 3; x=${x} + 1) {
|
||||
Background(demo-instruct);
|
||||
WaitExten();
|
||||
};
|
||||
};
|
||||
2 => {
|
||||
Background(demo-moreinfo);
|
||||
goto s|instructions;
|
||||
};
|
||||
3 => {
|
||||
Set(LANGUAGE()=fr);
|
||||
goto s|restart;
|
||||
};
|
||||
1000 => {
|
||||
goto ael-default|s|1;
|
||||
};
|
||||
500 => {
|
||||
Playback(demo-abouttotry);
|
||||
Dial(IAX2/guest@misery.digium.com/s@default);
|
||||
Playback(demo-nogo);
|
||||
goto s|instructions;
|
||||
};
|
||||
600 => {
|
||||
Playback(demo-echotest);
|
||||
Echo();
|
||||
Playback(demo-echodone);
|
||||
goto s|instructions;
|
||||
};
|
||||
_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
|
||||
8500 => {
|
||||
VoicemailMain();
|
||||
goto s|instructions;
|
||||
};
|
||||
# => {
|
||||
Playback(demo-thanks);
|
||||
Hangup();
|
||||
};
|
||||
t => goto #|1;
|
||||
i => Playback(invalid);
|
||||
};
|
||||
|
||||
|
||||
|
@ -383,9 +383,9 @@ context ael-default {
|
|||
// By default we include the demo. In a production system, you
|
||||
// probably don't want to have the demo there.
|
||||
|
||||
includes {
|
||||
ael-demo;
|
||||
};
|
||||
includes {
|
||||
ael-demo;
|
||||
};
|
||||
//
|
||||
// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
|
||||
// Note that you must have a [sipprovider] section in sip.conf whereas
|
||||
|
|
|
@ -430,7 +430,7 @@ exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
|
|||
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
|
||||
|
||||
exten => stdexten-BUSY,1,Voicemail(${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)
|
||||
|
||||
|
|
|
@ -97,103 +97,103 @@ TRUNKMSD = 1
|
|||
--
|
||||
|
||||
function outgoing_local(c, e)
|
||||
app.dial("DAHDI/1/" .. e, "", "")
|
||||
app.dial("DAHDI/1/" .. e, "", "")
|
||||
end
|
||||
|
||||
function demo_instruct()
|
||||
app.background("demo-instruct")
|
||||
app.waitexten()
|
||||
app.background("demo-instruct")
|
||||
app.waitexten()
|
||||
end
|
||||
|
||||
function demo_congrats()
|
||||
app.background("demo-congrats")
|
||||
demo_instruct()
|
||||
app.background("demo-congrats")
|
||||
demo_instruct()
|
||||
end
|
||||
|
||||
-- Answer the chanel and play the demo sound files
|
||||
function demo_start(context, exten)
|
||||
app.wait(1)
|
||||
app.answer()
|
||||
app.wait(1)
|
||||
app.answer()
|
||||
|
||||
channel.TIMEOUT("digit"):set(5)
|
||||
channel.TIMEOUT("response"):set(10)
|
||||
-- app.set("TIMEOUT(digit)=5")
|
||||
-- app.set("TIMEOUT(response)=10")
|
||||
channel.TIMEOUT("digit"):set(5)
|
||||
channel.TIMEOUT("response"):set(10)
|
||||
-- app.set("TIMEOUT(digit)=5")
|
||||
-- app.set("TIMEOUT(response)=10")
|
||||
|
||||
demo_congrats(context, exten)
|
||||
demo_congrats(context, exten)
|
||||
end
|
||||
|
||||
function demo_hangup()
|
||||
app.playback("demo-thanks")
|
||||
app.hangup()
|
||||
app.playback("demo-thanks")
|
||||
app.hangup()
|
||||
end
|
||||
|
||||
extensions = {
|
||||
demo = {
|
||||
s = demo_start;
|
||||
demo = {
|
||||
s = demo_start;
|
||||
|
||||
["2"] = function()
|
||||
app.background("demo-moreinfo")
|
||||
demo_instruct()
|
||||
end;
|
||||
["3"] = function ()
|
||||
channel.LANGUAGE():set("fr") -- set the language to french
|
||||
demo_congrats()
|
||||
end;
|
||||
["2"] = function()
|
||||
app.background("demo-moreinfo")
|
||||
demo_instruct()
|
||||
end;
|
||||
["3"] = function ()
|
||||
channel.LANGUAGE():set("fr") -- set the language to french
|
||||
demo_congrats()
|
||||
end;
|
||||
|
||||
["1000"] = function()
|
||||
app.goto("default", "s", 1)
|
||||
end;
|
||||
["1000"] = function()
|
||||
app.goto("default", "s", 1)
|
||||
end;
|
||||
|
||||
["1234"] = function()
|
||||
app.playback("transfer", "skip")
|
||||
-- do a dial here
|
||||
end;
|
||||
["1234"] = function()
|
||||
app.playback("transfer", "skip")
|
||||
-- do a dial here
|
||||
end;
|
||||
|
||||
["1235"] = function()
|
||||
app.voicemail("1234", "u")
|
||||
end;
|
||||
["1235"] = function()
|
||||
app.voicemail("1234", "u")
|
||||
end;
|
||||
|
||||
["1236"] = function()
|
||||
app.dial("Console/dsp")
|
||||
app.voicemail(1234, "b")
|
||||
end;
|
||||
["1236"] = function()
|
||||
app.dial("Console/dsp")
|
||||
app.voicemail(1234, "b")
|
||||
end;
|
||||
|
||||
["#"] = demo_hangup;
|
||||
t = demo_hangup;
|
||||
i = function()
|
||||
app.playback("invalid")
|
||||
demo_instruct()
|
||||
end;
|
||||
["#"] = demo_hangup;
|
||||
t = demo_hangup;
|
||||
i = function()
|
||||
app.playback("invalid")
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
["500"] = function()
|
||||
app.playback("demo-abouttotry")
|
||||
app.dial("IAX2/guest@misery.digium.com/s@default")
|
||||
app.playback("demo-nogo")
|
||||
demo_instruct()
|
||||
end;
|
||||
["500"] = function()
|
||||
app.playback("demo-abouttotry")
|
||||
app.dial("IAX2/guest@misery.digium.com/s@default")
|
||||
app.playback("demo-nogo")
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
["600"] = function()
|
||||
app.playback("demo-echotest")
|
||||
app.echo()
|
||||
app.playback("demo-echodone")
|
||||
demo_instruct()
|
||||
end;
|
||||
["600"] = function()
|
||||
app.playback("demo-echotest")
|
||||
app.echo()
|
||||
app.playback("demo-echodone")
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
["8500"] = function()
|
||||
app.voicemailmain()
|
||||
demo_instruct()
|
||||
end;
|
||||
["8500"] = function()
|
||||
app.voicemailmain()
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
};
|
||||
};
|
||||
|
||||
default = {
|
||||
-- by default, do the demo
|
||||
include = {"demo"};
|
||||
};
|
||||
default = {
|
||||
-- by default, do the demo
|
||||
include = {"demo"};
|
||||
};
|
||||
|
||||
["local"] = {
|
||||
["_NXXXXXX"] = outgoing_local;
|
||||
};
|
||||
["local"] = {
|
||||
["_NXXXXXX"] = outgoing_local;
|
||||
};
|
||||
}
|
||||
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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.
|
||||
;
|
||||
|
|
|
@ -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".
|
||||
;-----------------------------------------------------------------------------------
|
||||
|
|
|
@ -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.
|
||||
|
|
|
@ -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.
|
||||
|
|
|
@ -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.
|
||||
;
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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]
|
||||
|
|
|
@ -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".
|
||||
;-----------------------------------------------------------------------------------
|
||||
|
|
|
@ -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:
|
||||
|
|
|
@ -3,75 +3,75 @@
|
|||
;
|
||||
|
||||
[general]
|
||||
; General config options, with default values shown.
|
||||
; You should use one section per device, with [general] being used
|
||||
; for the first device and also as a template for other devices.
|
||||
;
|
||||
; All but 'debug' can go also in the device-specific sections.
|
||||
;
|
||||
; debug = 0x0 ; misc debug flags, default is 0
|
||||
; General config options, with default values shown.
|
||||
; You should use one section per device, with [general] being used
|
||||
; for the first device and also as a template for other devices.
|
||||
;
|
||||
; All but 'debug' can go also in the device-specific sections.
|
||||
;
|
||||
; debug = 0x0 ; misc debug flags, default is 0
|
||||
|
||||
; Set the device to use for I/O
|
||||
; device = /dev/dsp
|
||||
; Set the device to use for I/O
|
||||
; device = /dev/dsp
|
||||
|
||||
; Optional mixer command to run upon startup (e.g. to set
|
||||
; volume levels, mutes, etc.
|
||||
; mixer =
|
||||
; Optional mixer command to run upon startup (e.g. to set
|
||||
; volume levels, mutes, etc.
|
||||
; mixer =
|
||||
|
||||
; Software mic volume booster (or attenuator), useful for sound
|
||||
; cards or microphones with poor sensitivity. The volume level
|
||||
; is in dB, ranging from -20.0 to +20.0
|
||||
; boost = n ; mic volume boost in dB
|
||||
; Software mic volume booster (or attenuator), useful for sound
|
||||
; cards or microphones with poor sensitivity. The volume level
|
||||
; is in dB, ranging from -20.0 to +20.0
|
||||
; boost = n ; mic volume boost in dB
|
||||
|
||||
; Set the callerid for outgoing calls
|
||||
; callerid = John Doe <555-1234>
|
||||
; Set the callerid for outgoing calls
|
||||
; callerid = John Doe <555-1234>
|
||||
|
||||
; autoanswer = no ; no autoanswer on call
|
||||
; autohangup = yes ; hangup when other party closes
|
||||
; extension = s ; default extension to call
|
||||
; context = default ; default context for outgoing calls
|
||||
; language = "" ; default language
|
||||
; autoanswer = no ; no autoanswer on call
|
||||
; autohangup = yes ; hangup when other party closes
|
||||
; extension = s ; default extension to call
|
||||
; context = default ; default context for outgoing calls
|
||||
; language = "" ; default language
|
||||
|
||||
; If you set overridecontext to 'yes', then the whole dial string
|
||||
; will be interpreted as an extension, which is extremely useful
|
||||
; to dial SIP, IAX and other extensions which use the '@' character.
|
||||
; The default is 'no' just for backward compatibility, but the
|
||||
; suggestion is to change it.
|
||||
; overridecontext = no ; if 'no', the last @ will start the context
|
||||
; if 'yes' the whole string is an extension.
|
||||
; If you set overridecontext to 'yes', then the whole dial string
|
||||
; will be interpreted as an extension, which is extremely useful
|
||||
; to dial SIP, IAX and other extensions which use the '@' character.
|
||||
; The default is 'no' just for backward compatibility, but the
|
||||
; suggestion is to change it.
|
||||
; overridecontext = no ; if 'no', the last @ will start the context
|
||||
; if 'yes' the whole string is an extension.
|
||||
|
||||
; low level device parameters in case you have problems with the
|
||||
; device driver on your operating system. You should not touch these
|
||||
; unless you know what you are doing.
|
||||
; queuesize = 10 ; frames in device driver
|
||||
; frags = 8 ; argument to SETFRAGMENT
|
||||
; low level device parameters in case you have problems with the
|
||||
; device driver on your operating system. You should not touch these
|
||||
; unless you know what you are doing.
|
||||
; queuesize = 10 ; frames in device driver
|
||||
; frags = 8 ; argument to SETFRAGMENT
|
||||
|
||||
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
|
||||
; OSS channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The OSS channel can't accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive OSS side will always
|
||||
; be used if the sending side can create jitter.
|
||||
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
|
||||
; OSS channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The OSS channel can't accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive OSS side will always
|
||||
; be used if the sending side can create jitter.
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
;-----------------------------------------------------------------------------------
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
;-----------------------------------------------------------------------------------
|
||||
|
||||
; below is an entry for a second console channel
|
||||
; [card1]
|
||||
; device = /dev/dsp1 ; alternate device
|
||||
; device = /dev/dsp1 ; alternate device
|
||||
|
||||
; Below are the settings to support video. You can include them
|
||||
; in your general configuration as [general](+,video)
|
||||
|
@ -79,26 +79,26 @@
|
|||
; Section names used here are only examples.
|
||||
|
||||
[my_video](!) ; you can just include in your config
|
||||
videodevice = /dev/video0 ; uses your V4L webcam as video source
|
||||
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
|
||||
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
|
||||
videodevice = /dev/video0 ; uses your V4L webcam as video source
|
||||
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
|
||||
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
|
||||
|
||||
; video_size is the geometry used by the encoder.
|
||||
; Depending on the codec your choice is restricted.
|
||||
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
|
||||
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
|
||||
; video_size is the geometry used by the encoder.
|
||||
; Depending on the codec your choice is restricted.
|
||||
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
|
||||
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
|
||||
|
||||
; You can also set the geometry used for the camera, local display and remote display.
|
||||
; The local window is on the right, the remote window is on the left.
|
||||
; Right clicking with the mouse on a video window increases the size,
|
||||
; center-clicking reduces the size.
|
||||
camera_size = cif
|
||||
remote_size = cif
|
||||
local_size = qcif
|
||||
; You can also set the geometry used for the camera, local display and remote display.
|
||||
; The local window is on the right, the remote window is on the left.
|
||||
; Right clicking with the mouse on a video window increases the size,
|
||||
; center-clicking reduces the size.
|
||||
camera_size = cif
|
||||
remote_size = cif
|
||||
local_size = qcif
|
||||
|
||||
bitrate = 60000 ; rate told to ffmpeg.
|
||||
fps = 5 ; frames per second from the source.
|
||||
; qmin = 3 ; quantizer value passed to the encoder.
|
||||
bitrate = 60000 ; rate told to ffmpeg.
|
||||
fps = 5 ; frames per second from the source.
|
||||
; qmin = 3 ; quantizer value passed to the encoder.
|
||||
|
||||
; The keypad is made of an image (in any format supported by SDL_image)
|
||||
; and some configuration entries indicating the location and function of buttons.
|
||||
|
@ -115,30 +115,30 @@ fps = 5 ; frames per second from the source.
|
|||
; diameter of the ellipse.
|
||||
;
|
||||
[my_skin](!)
|
||||
keypad = /tmp/keypad.jpg
|
||||
region = 1 rect 19 18 67 18 28
|
||||
region = 2 rect 84 18 133 18 28
|
||||
region = 3 rect 152 18 201 18 28
|
||||
region = 4 rect 19 60 67 60 28
|
||||
region = 5 rect 84 60 133 60 28
|
||||
region = 6 rect 152 60 201 60 28
|
||||
region = 7 rect 19 103 67 103 28
|
||||
region = 8 rect 84 103 133 103 28
|
||||
region = 9 rect 152 103 201 103 28
|
||||
region = * rect 19 146 67 146 28
|
||||
region = 0 rect 84 146 133 146 28
|
||||
region = # rect 152 146 201 146 28
|
||||
region = pickup rect 229 15 267 15 40
|
||||
region = hangup rect 230 66 270 64 40
|
||||
region = mute circle 232 141 264 141 33
|
||||
region = sendvideo circle 235 185 266 185 33
|
||||
region = autoanswer rect 228 212 275 212 50
|
||||
keypad = /tmp/keypad.jpg
|
||||
region = 1 rect 19 18 67 18 28
|
||||
region = 2 rect 84 18 133 18 28
|
||||
region = 3 rect 152 18 201 18 28
|
||||
region = 4 rect 19 60 67 60 28
|
||||
region = 5 rect 84 60 133 60 28
|
||||
region = 6 rect 152 60 201 60 28
|
||||
region = 7 rect 19 103 67 103 28
|
||||
region = 8 rect 84 103 133 103 28
|
||||
region = 9 rect 152 103 201 103 28
|
||||
region = * rect 19 146 67 146 28
|
||||
region = 0 rect 84 146 133 146 28
|
||||
region = # rect 152 146 201 146 28
|
||||
region = pickup rect 229 15 267 15 40
|
||||
region = hangup rect 230 66 270 64 40
|
||||
region = mute circle 232 141 264 141 33
|
||||
region = sendvideo circle 235 185 266 185 33
|
||||
region = autoanswer rect 228 212 275 212 50
|
||||
|
||||
; another skin with entries for the keypad and a small font
|
||||
; to write to the message boards in the skin.
|
||||
[skin2](!)
|
||||
keypad = /tmp/kpad2.jpg
|
||||
keypad_font = /tmp/font.png
|
||||
keypad = /tmp/kpad2.jpg
|
||||
keypad_font = /tmp/font.png
|
||||
|
||||
; to add video support, uncomment this and remember to install
|
||||
; the keypad and keypad_font files to the right place
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
||||
|
|
|
@ -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.
|
||||
|
|
|
@ -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]
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -4,8 +4,8 @@
|
|||
|
||||
[general]
|
||||
mode=old ; method for playing numbers and dates
|
||||
; old - using asterisk core function
|
||||
; new - using this configuration file
|
||||
; old - using asterisk core function
|
||||
; new - using this configuration file
|
||||
|
||||
; The new language routines produce strings of the form
|
||||
; prefix:[format:]data
|
||||
|
@ -75,126 +75,126 @@ mode=old ; method for playing numbers and dates
|
|||
; language-independent
|
||||
|
||||
[digit-base](!) ; base rule for digit strings
|
||||
; XXX incomplete yet
|
||||
_digit:[0-9] => digits/${SAY}
|
||||
_digit:[-] => letters/dash
|
||||
_digit:[*] => letters/star
|
||||
_digit:[@] => letters/at
|
||||
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
|
||||
; XXX incomplete yet
|
||||
_digit:[0-9] => digits/${SAY}
|
||||
_digit:[-] => letters/dash
|
||||
_digit:[*] => letters/star
|
||||
_digit:[@] => letters/at
|
||||
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
|
||||
|
||||
[date-base](!) ; base rules for dates and times
|
||||
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
|
||||
; these rule map the strftime attributes.
|
||||
_date:Y:. => num:${SAY:0:4} ; year, 19xx
|
||||
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
|
||||
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
|
||||
_date:[de]:. => num:${SAY:6:2} ; day of month
|
||||
_date:[hH]:. => num:${SAY:8:2} ; hour
|
||||
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
|
||||
_date:[M]:. => num:${SAY:10:2} ; minute
|
||||
; XXX too bad the '?' function does not remove the quotes
|
||||
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
|
||||
_date:[pP]:. => digits/p-m ; am pm
|
||||
_date:[S]:. => num:${SAY:13:2} ; seconds
|
||||
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
|
||||
; these rule map the strftime attributes.
|
||||
_date:Y:. => num:${SAY:0:4} ; year, 19xx
|
||||
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
|
||||
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
|
||||
_date:[de]:. => num:${SAY:6:2} ; day of month
|
||||
_date:[hH]:. => num:${SAY:8:2} ; hour
|
||||
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
|
||||
_date:[M]:. => num:${SAY:10:2} ; minute
|
||||
; XXX too bad the '?' function does not remove the quotes
|
||||
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
|
||||
_date:[pP]:. => digits/p-m ; am pm
|
||||
_date:[S]:. => num:${SAY:13:2} ; seconds
|
||||
|
||||
[en-base](!)
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1X => digits/${SAY}
|
||||
_[n]um:[2-9]0 => digits/${SAY}
|
||||
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
|
||||
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1X => digits/${SAY}
|
||||
_[n]um:[2-9]0 => digits/${SAY}
|
||||
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
|
||||
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
|
||||
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
|
||||
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
|
||||
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
|
||||
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
|
||||
|
||||
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
|
||||
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
|
||||
|
||||
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
|
||||
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
|
||||
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
|
||||
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
|
||||
|
||||
; enumeration
|
||||
_e[n]um:X => digits/h-${SAY}
|
||||
_e[n]um:1X => digits/h-${SAY}
|
||||
_e[n]um:[2-9]0 => digits/h-${SAY}
|
||||
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
|
||||
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
|
||||
; enumeration
|
||||
_e[n]um:X => digits/h-${SAY}
|
||||
_e[n]um:1X => digits/h-${SAY}
|
||||
_e[n]um:[2-9]0 => digits/h-${SAY}
|
||||
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
|
||||
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
|
||||
|
||||
[it](digit-base,date-base)
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1X => digits/${SAY}
|
||||
_[n]um:[2-9]0 => digits/${SAY}
|
||||
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
|
||||
_[n]um:1XX => digits/hundred, num:${SAY:1}
|
||||
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1X => digits/${SAY}
|
||||
_[n]um:[2-9]0 => digits/${SAY}
|
||||
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
|
||||
_[n]um:1XX => digits/hundred, num:${SAY:1}
|
||||
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
|
||||
_[n]um:1XXX => digits/thousand, num:${SAY:1}
|
||||
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
|
||||
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
|
||||
_[n]um:1XXX => digits/thousand, num:${SAY:1}
|
||||
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
|
||||
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
|
||||
|
||||
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
|
||||
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
|
||||
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
|
||||
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
|
||||
|
||||
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
|
||||
_date::. => date:AdBY:${SAY}
|
||||
_time::. => date:IMp:${SAY}
|
||||
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
|
||||
_date::. => date:AdBY:${SAY}
|
||||
_time::. => date:IMp:${SAY}
|
||||
|
||||
[en](en-base,date-base,digit-base)
|
||||
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
|
||||
_date::. => date:AdBY:${SAY}
|
||||
_time::. => date:IMp:${SAY}
|
||||
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
|
||||
_date::. => date:AdBY:${SAY}
|
||||
_time::. => date:IMp:${SAY}
|
||||
|
||||
[de](date-base,digit-base)
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1X => digits/${SAY}
|
||||
_[n]um:[2-9]0 => digits/${SAY}
|
||||
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
|
||||
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
|
||||
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
|
||||
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
|
||||
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
|
||||
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
|
||||
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1X => digits/${SAY}
|
||||
_[n]um:[2-9]0 => digits/${SAY}
|
||||
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
|
||||
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
|
||||
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
|
||||
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
|
||||
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
|
||||
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
|
||||
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
|
||||
|
||||
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
|
||||
_date::. => date:AdBY:${SAY}
|
||||
_time::. => date:IMp:${SAY}
|
||||
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
|
||||
_date::. => date:AdBY:${SAY}
|
||||
_time::. => date:IMp:${SAY}
|
||||
|
||||
[hu](digit-base,date-base)
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1[1-9] => digits/10en, digits/${SAY:1}
|
||||
_[n]um:2[1-9] => digits/20on, digits/${SAY:1}
|
||||
_[n]um:[1-9]0 => digits/${SAY}
|
||||
_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
|
||||
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
_[n]um:0. => num:${SAY:1}
|
||||
_[n]um:X => digits/${SAY}
|
||||
_[n]um:1[1-9] => digits/10en, digits/${SAY:1}
|
||||
_[n]um:2[1-9] => digits/20on, digits/${SAY:1}
|
||||
_[n]um:[1-9]0 => digits/${SAY}
|
||||
_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
|
||||
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
|
||||
|
||||
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
|
||||
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
|
||||
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
|
||||
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
|
||||
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
|
||||
|
||||
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
|
||||
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
|
||||
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
|
||||
|
||||
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
|
||||
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
|
||||
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
|
||||
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
|
||||
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
|
||||
|
||||
_datetime::. => date:YBdA k 'ora' M 'perc':${SAY}
|
||||
_date::. => date:YBdA:${SAY}
|
||||
_time::. => date:k 'ora' M 'perc':${SAY}
|
||||
_datetime::. => date:YBdA k 'ora' M 'perc':${SAY}
|
||||
_date::. => date:YBdA:${SAY}
|
||||
_time::. => date:k 'ora' M 'perc':${SAY}
|
||||
|
|
|
@ -88,18 +88,18 @@
|
|||
context=default ; Default context for incoming calls
|
||||
;allowguest=no ; Allow or reject guest calls (default is yes)
|
||||
;match_auth_username=yes ; if available, match user entry using the
|
||||
; 'username' field from the authentication line
|
||||
; instead of the From: field.
|
||||
; 'username' field from the authentication line
|
||||
; instead of the From: field.
|
||||
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
|
||||
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
|
||||
; Default is enabled
|
||||
; Default is enabled
|
||||
;realm=mydomain.tld ; Realm for digest authentication
|
||||
; defaults to "asterisk". If you set a system name in
|
||||
; asterisk.conf, it defaults to that system name
|
||||
; Realms MUST be globally unique according to RFC 3261
|
||||
; Set this to your host name or domain name
|
||||
; defaults to "asterisk". If you set a system name in
|
||||
; asterisk.conf, it defaults to that system name
|
||||
; Realms MUST be globally unique according to RFC 3261
|
||||
; Set this to your host name or domain name
|
||||
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
|
||||
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
||||
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
||||
|
||||
;
|
||||
; Note that the TCP and TLS support for chan_sip is currently considered
|
||||
|
@ -109,20 +109,20 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0
|
|||
;
|
||||
tcpenable=no ; Enable server for incoming TCP connections (default is no)
|
||||
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
|
||||
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
||||
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
||||
|
||||
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
|
||||
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
|
||||
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
|
||||
; Remember that the IP address must match the common name (hostname) in the
|
||||
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
|
||||
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
|
||||
; Remember that the IP address must match the common name (hostname) in the
|
||||
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
|
||||
|
||||
;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
|
||||
; default is to look for "asterisk.pem" in current directory
|
||||
; default is to look for "asterisk.pem" in current directory
|
||||
|
||||
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
|
||||
; If no tlsprivatekey is specified, tlscertfile is searched for
|
||||
; for both public and private key.
|
||||
; If no tlsprivatekey is specified, tlscertfile is searched for
|
||||
; for both public and private key.
|
||||
|
||||
;tlscafile=</path/to/certificate>
|
||||
; If the server your connecting to uses a self signed certificate
|
||||
|
@ -146,20 +146,20 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
|
|||
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
|
||||
;
|
||||
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
|
||||
; Specify protocol for outbound client connections.
|
||||
; If left unspecified, the default is sslv2.
|
||||
; Specify protocol for outbound client connections.
|
||||
; If left unspecified, the default is sslv2.
|
||||
|
||||
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
; Note: Asterisk only uses the first host
|
||||
; in SRV records
|
||||
; Disabling DNS SRV lookups disables the
|
||||
; ability to place SIP calls based on domain
|
||||
; names to some other SIP users on the Internet
|
||||
; Note: Asterisk only uses the first host
|
||||
; in SRV records
|
||||
; Disabling DNS SRV lookups disables the
|
||||
; ability to place SIP calls based on domain
|
||||
; names to some other SIP users on the Internet
|
||||
|
||||
;pedantic=yes ; Enable checking of tags in headers,
|
||||
; international character conversions in URIs
|
||||
; and multiline formatted headers for strict
|
||||
; SIP compatibility (defaults to "no")
|
||||
; international character conversions in URIs
|
||||
; and multiline formatted headers for strict
|
||||
; SIP compatibility (defaults to "no")
|
||||
|
||||
; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
|
||||
;tos_sip=cs3 ; Sets TOS for SIP packets.
|
||||
|
@ -173,32 +173,32 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;cos_text=3 ; Sets 802.1p priority for RTP text packets.
|
||||
|
||||
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
|
||||
; and subscriptions (seconds)
|
||||
; and subscriptions (seconds)
|
||||
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
|
||||
;defaultexpiry=120 ; Default length of incoming/outgoing registration
|
||||
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
|
||||
;qualifyfreq=60 ; Qualification: How often to check for the
|
||||
; host to be up in seconds
|
||||
; Set to low value if you use low timeout for
|
||||
; NAT of UDP sessions
|
||||
; host to be up in seconds
|
||||
; Set to low value if you use low timeout for
|
||||
; NAT of UDP sessions
|
||||
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
|
||||
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
|
||||
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
|
||||
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
|
||||
; fully. Enable this option to not get error messages
|
||||
; when sending MWI to phones with this bug.
|
||||
; fully. Enable this option to not get error messages
|
||||
; when sending MWI to phones with this bug.
|
||||
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
|
||||
; the From: header as the "name" portion. Also fill the
|
||||
; "user" portion of the URI in the From: header with this
|
||||
; value if no fromuser is set
|
||||
; Default: empty
|
||||
; the From: header as the "name" portion. Also fill the
|
||||
; "user" portion of the URI in the From: header with this
|
||||
; value if no fromuser is set
|
||||
; Default: empty
|
||||
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
|
||||
; Message-Account in the MWI notify message
|
||||
; defaults to "asterisk"
|
||||
; Message-Account in the MWI notify message
|
||||
; defaults to "asterisk"
|
||||
|
||||
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
|
||||
; rather than advertising all joint codec capabilities. This
|
||||
; limits the other side's codec choice to exactly what we prefer.
|
||||
; rather than advertising all joint codec capabilities. This
|
||||
; limits the other side's codec choice to exactly what we prefer.
|
||||
|
||||
;disallow=all ; First disallow all codecs
|
||||
;allow=ulaw ; Allow codecs in order of preference
|
||||
|
@ -220,83 +220,83 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;mohsuggest=default
|
||||
;
|
||||
;parkinglot=plaza ; Sets the default parking lot for call parking
|
||||
; This may also be set for individual users/peers
|
||||
; Parkinglots are configured in features.conf
|
||||
; This may also be set for individual users/peers
|
||||
; Parkinglots are configured in features.conf
|
||||
;language=en ; Default language setting for all users/peers
|
||||
; This may also be set for individual users/peers
|
||||
; This may also be set for individual users/peers
|
||||
;relaxdtmf=yes ; Relax dtmf handling
|
||||
;trustrpid = no ; If Remote-Party-ID should be trusted
|
||||
;sendrpid = yes ; If Remote-Party-ID should be sent
|
||||
;sendrpid = rpid ; Use the "Remote-Party-ID" header
|
||||
; to send the identity of the remote party
|
||||
; This is identical to sendrpid=yes
|
||||
; to send the identity of the remote party
|
||||
; This is identical to sendrpid=yes
|
||||
;sendrpid = pai ; Use the "P-Asserted-Identity" header
|
||||
; to send the identity of the remote party
|
||||
; to send the identity of the remote party
|
||||
;rpid_update = no ; In certain cases, the only method by which a connected line
|
||||
; change may be immediately transmitted is with a SIP UPDATE request.
|
||||
; If communicating with another Asterisk server, and you wish to be able
|
||||
; transmit such UPDATE messages to it, then you must enable this option.
|
||||
; Otherwise, we will have to wait until we can send a reinvite to
|
||||
; transmit the information.
|
||||
; change may be immediately transmitted is with a SIP UPDATE request.
|
||||
; If communicating with another Asterisk server, and you wish to be able
|
||||
; transmit such UPDATE messages to it, then you must enable this option.
|
||||
; Otherwise, we will have to wait until we can send a reinvite to
|
||||
; transmit the information.
|
||||
|
||||
;progressinband=never ; If we should generate in-band ringing always
|
||||
; use 'never' to never use in-band signalling, even in cases
|
||||
; where some buggy devices might not render it
|
||||
; Valid values: yes, no, never Default: never
|
||||
; use 'never' to never use in-band signalling, even in cases
|
||||
; where some buggy devices might not render it
|
||||
; Valid values: yes, no, never Default: never
|
||||
;useragent=Asterisk PBX ; Allows you to change the user agent string
|
||||
; The default user agent string also contains the Asterisk
|
||||
; version. If you don't want to expose this, change the
|
||||
; useragent string.
|
||||
; The default user agent string also contains the Asterisk
|
||||
; version. If you don't want to expose this, change the
|
||||
; useragent string.
|
||||
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
|
||||
; Like the useragent parameter, the default user agent string
|
||||
; also contains the Asterisk version.
|
||||
; Like the useragent parameter, the default user agent string
|
||||
; also contains the Asterisk version.
|
||||
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
|
||||
; This field MUST NOT contain spaces
|
||||
; This field MUST NOT contain spaces
|
||||
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
|
||||
; Note that promiscredir when redirects are made to the
|
||||
; local system will cause loops since Asterisk is incapable
|
||||
; of performing a "hairpin" call.
|
||||
; Note that promiscredir when redirects are made to the
|
||||
; local system will cause loops since Asterisk is incapable
|
||||
; of performing a "hairpin" call.
|
||||
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
|
||||
; a valid phone number
|
||||
; a valid phone number
|
||||
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
|
||||
; Other options:
|
||||
; info : SIP INFO messages (application/dtmf-relay)
|
||||
; shortinfo : SIP INFO messages (application/dtmf)
|
||||
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
|
||||
; auto : Use rfc2833 if offered, inband otherwise
|
||||
; Other options:
|
||||
; info : SIP INFO messages (application/dtmf-relay)
|
||||
; shortinfo : SIP INFO messages (application/dtmf)
|
||||
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
|
||||
; auto : Use rfc2833 if offered, inband otherwise
|
||||
|
||||
;compactheaders = yes ; send compact sip headers.
|
||||
;
|
||||
;videosupport=yes ; Turn on support for SIP video. You need to turn this
|
||||
; on in this section to get any video support at all.
|
||||
; You can turn it off on a per peer basis if the general
|
||||
; video support is enabled, but you can't enable it for
|
||||
; one peer only without enabling in the general section.
|
||||
; If you set videosupport to "always", then RTP ports will
|
||||
; always be set up for video, even on clients that don't
|
||||
; support it. This assists callfile-derived calls and
|
||||
; certain transferred calls to use always use video when
|
||||
; available. [yes|NO|always]
|
||||
; on in this section to get any video support at all.
|
||||
; You can turn it off on a per peer basis if the general
|
||||
; video support is enabled, but you can't enable it for
|
||||
; one peer only without enabling in the general section.
|
||||
; If you set videosupport to "always", then RTP ports will
|
||||
; always be set up for video, even on clients that don't
|
||||
; support it. This assists callfile-derived calls and
|
||||
; certain transferred calls to use always use video when
|
||||
; available. [yes|NO|always]
|
||||
|
||||
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
|
||||
; Videosupport and maxcallbitrate is settable
|
||||
; for peers and users as well
|
||||
; Videosupport and maxcallbitrate is settable
|
||||
; for peers and users as well
|
||||
;callevents=no ; generate manager events when sip ua
|
||||
; performs events (e.g. hold)
|
||||
; performs events (e.g. hold)
|
||||
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
|
||||
; authenticate with Asterisk. Peerstatus will be "rejected".
|
||||
; authenticate with Asterisk. Peerstatus will be "rejected".
|
||||
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
|
||||
; for any reason, always reject with an identical response
|
||||
; equivalent to valid username and invalid password/hash
|
||||
; instead of letting the requester know whether there was
|
||||
; a matching user or peer for their request. This reduces
|
||||
; the ability of an attacker to scan for valid SIP usernames.
|
||||
; for any reason, always reject with an identical response
|
||||
; equivalent to valid username and invalid password/hash
|
||||
; instead of letting the requester know whether there was
|
||||
; a matching user or peer for their request. This reduces
|
||||
; the ability of an attacker to scan for valid SIP usernames.
|
||||
|
||||
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
|
||||
; order instead of RFC3551 packing order (this is required
|
||||
; for Sipura and Grandstream ATAs, among others). This is
|
||||
; contrary to the RFC3551 specification, the peer _should_
|
||||
; be negotiating AAL2-G726-32 instead :-(
|
||||
; order instead of RFC3551 packing order (this is required
|
||||
; for Sipura and Grandstream ATAs, among others). This is
|
||||
; contrary to the RFC3551 specification, the peer _should_
|
||||
; be negotiating AAL2-G726-32 instead :-(
|
||||
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
|
||||
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
|
||||
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
|
||||
|
@ -304,18 +304,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; ; (could also be tcp,udp) - defining transports on the proxy line only
|
||||
; ; applies for the global proxy, otherwise use the transport= option
|
||||
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
|
||||
; your localnet setting. Unless you have some sort of strange network
|
||||
; setup you will not need to enable this.
|
||||
; your localnet setting. Unless you have some sort of strange network
|
||||
; setup you will not need to enable this.
|
||||
|
||||
;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
|
||||
; as any IP address used for staticly defined
|
||||
; hosts. This helps avoid the configuration
|
||||
; error of allowing your users to register at
|
||||
; the same address as a SIP provider.
|
||||
; as any IP address used for staticly defined
|
||||
; hosts. This helps avoid the configuration
|
||||
; error of allowing your users to register at
|
||||
; the same address as a SIP provider.
|
||||
|
||||
;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
|
||||
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
|
||||
; register their phones.
|
||||
; register their phones.
|
||||
|
||||
;engine=asterisk ; RTP engine to use when communicating with the device
|
||||
|
||||
|
@ -332,9 +332,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;
|
||||
;regcontext=sipregistrations
|
||||
;regextenonqualify=yes ; Default "no"
|
||||
; If you have qualify on and the peer becomes unreachable
|
||||
; this setting will enforce inactivation of the regexten
|
||||
; extension for the peer
|
||||
; If you have qualify on and the peer becomes unreachable
|
||||
; this setting will enforce inactivation of the regexten
|
||||
; extension for the peer
|
||||
;
|
||||
;--------------------------- SIP timers ----------------------------------------------------
|
||||
; These timers are used primarily in INVITE transactions.
|
||||
|
@ -342,13 +342,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; Asterisk and the device if you have qualify=yes for the device.
|
||||
;
|
||||
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
|
||||
; Defaults to 100 ms
|
||||
; Defaults to 100 ms
|
||||
;timert1=500 ; Default T1 timer
|
||||
; Defaults to 500 ms or the measured round-trip
|
||||
; time to a peer (qualify=yes).
|
||||
; Defaults to 500 ms or the measured round-trip
|
||||
; time to a peer (qualify=yes).
|
||||
;timerb=32000 ; Call setup timer. If a provisional response is not received
|
||||
; in this amount of time, the call will autocongest
|
||||
; Defaults to 64*timert1
|
||||
; in this amount of time, the call will autocongest
|
||||
; Defaults to 64*timert1
|
||||
|
||||
;--------------------------- RTP timers ----------------------------------------------------
|
||||
; These timers are currently used for both audio and video streams. The RTP timeouts
|
||||
|
@ -356,15 +356,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; The settings are settable in the global section as well as per device
|
||||
;
|
||||
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
|
||||
; on the audio channel
|
||||
; when we're not on hold. This is to be able to hangup
|
||||
; a call in the case of a phone disappearing from the net,
|
||||
; like a powerloss or grandma tripping over a cable.
|
||||
; on the audio channel
|
||||
; when we're not on hold. This is to be able to hangup
|
||||
; a call in the case of a phone disappearing from the net,
|
||||
; like a powerloss or grandma tripping over a cable.
|
||||
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
|
||||
; on the audio channel
|
||||
; when we're on hold (must be > rtptimeout)
|
||||
; on the audio channel
|
||||
; when we're on hold (must be > rtptimeout)
|
||||
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
|
||||
; (default is off - zero)
|
||||
; (default is off - zero)
|
||||
|
||||
;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
|
||||
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
|
||||
|
@ -403,11 +403,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
|
||||
;--------------------------- SIP DEBUGGING ---------------------------------------------------
|
||||
;sipdebug = yes ; Turn on SIP debugging by default, from
|
||||
; the moment the channel loads this configuration
|
||||
; the moment the channel loads this configuration
|
||||
;recordhistory=yes ; Record SIP history by default
|
||||
; (see sip history / sip no history)
|
||||
; (see sip history / sip no history)
|
||||
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
|
||||
; SIP history is output to the DEBUG logging channel
|
||||
; SIP history is output to the DEBUG logging channel
|
||||
|
||||
|
||||
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
|
||||
|
@ -430,30 +430,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;
|
||||
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
|
||||
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
|
||||
; Useful to limit subscriptions to local extensions
|
||||
; Settable per peer/user also
|
||||
; Useful to limit subscriptions to local extensions
|
||||
; Settable per peer/user also
|
||||
;notifyringing = no ; Control whether subscriptions already INUSE get sent
|
||||
; RINGING when another call is sent (default: yes)
|
||||
; RINGING when another call is sent (default: yes)
|
||||
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
|
||||
; Turning on notifyringing and notifyhold will add a lot
|
||||
; more database transactions if you are using realtime.
|
||||
; Turning on notifyringing and notifyhold will add a lot
|
||||
; more database transactions if you are using realtime.
|
||||
;notifycid = yes ; Control whether caller ID information is sent along with
|
||||
; dialog-info+xml notifications (supported by snom phones).
|
||||
; Note that this feature will only work properly when the
|
||||
; incoming call is using the same extension and context that
|
||||
; is being used as the hint for the called extension. This means
|
||||
; that it won't work when using subscribecontext for your sip
|
||||
; user or peer (if subscribecontext is different than context).
|
||||
; This is also limited to a single caller, meaning that if an
|
||||
; extension is ringing because multiple calls are incoming,
|
||||
; only one will be used as the source of caller ID. Specify
|
||||
; 'ignore-context' to ignore the called context when looking
|
||||
; for the caller's channel. The default value is 'no.' Setting
|
||||
; notifycid to 'ignore-context' also causes call-pickups attempted
|
||||
; via SNOM's NOTIFY mechanism to set the context for the call pickup
|
||||
; to PICKUPMARK.
|
||||
; dialog-info+xml notifications (supported by snom phones).
|
||||
; Note that this feature will only work properly when the
|
||||
; incoming call is using the same extension and context that
|
||||
; is being used as the hint for the called extension. This means
|
||||
; that it won't work when using subscribecontext for your sip
|
||||
; user or peer (if subscribecontext is different than context).
|
||||
; This is also limited to a single caller, meaning that if an
|
||||
; extension is ringing because multiple calls are incoming,
|
||||
; only one will be used as the source of caller ID. Specify
|
||||
; 'ignore-context' to ignore the called context when looking
|
||||
; for the caller's channel. The default value is 'no.' Setting
|
||||
; notifycid to 'ignore-context' also causes call-pickups attempted
|
||||
; via SNOM's NOTIFY mechanism to set the context for the call pickup
|
||||
; to PICKUPMARK.
|
||||
;callcounter = yes ; Enable call counters on devices. This can be set per
|
||||
; device too.
|
||||
; device too.
|
||||
|
||||
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
|
||||
;
|
||||
|
@ -536,9 +536,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
|
||||
;registertimeout=20 ; retry registration calls every 20 seconds (default)
|
||||
;registerattempts=10 ; Number of registration attempts before we give up
|
||||
; 0 = continue forever, hammering the other server
|
||||
; until it accepts the registration
|
||||
; Default is 0 tries, continue forever
|
||||
; 0 = continue forever, hammering the other server
|
||||
; until it accepts the registration
|
||||
; Default is 0 tries, continue forever
|
||||
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
|
||||
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
|
||||
; by other phones.
|
||||
|
@ -645,43 +645,43 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
|
||||
;
|
||||
;canreinvite=yes ; Asterisk by default tries to redirect the
|
||||
; RTP media stream (audio) to go directly from
|
||||
; the caller to the callee. Some devices do not
|
||||
; support this (especially if one of them is behind a NAT).
|
||||
; The default setting is YES. If you have all clients
|
||||
; behind a NAT, or for some other reason wants Asterisk to
|
||||
; stay in the audio path, you may want to turn this off.
|
||||
; RTP media stream (audio) to go directly from
|
||||
; the caller to the callee. Some devices do not
|
||||
; support this (especially if one of them is behind a NAT).
|
||||
; The default setting is YES. If you have all clients
|
||||
; behind a NAT, or for some other reason wants Asterisk to
|
||||
; stay in the audio path, you may want to turn this off.
|
||||
|
||||
; This setting also affect direct RTP
|
||||
; at call setup (a new feature in 1.4 - setting up the
|
||||
; call directly between the endpoints instead of sending
|
||||
; a re-INVITE).
|
||||
; This setting also affect direct RTP
|
||||
; at call setup (a new feature in 1.4 - setting up the
|
||||
; call directly between the endpoints instead of sending
|
||||
; a re-INVITE).
|
||||
|
||||
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
|
||||
; the call directly with media peer-2-peer without re-invites.
|
||||
; Will not work for video and cases where the callee sends
|
||||
; RTP payloads and fmtp headers in the 200 OK that does not match the
|
||||
; callers INVITE. This will also fail if canreinvite is enabled when
|
||||
; the device is actually behind NAT.
|
||||
; the call directly with media peer-2-peer without re-invites.
|
||||
; Will not work for video and cases where the callee sends
|
||||
; RTP payloads and fmtp headers in the 200 OK that does not match the
|
||||
; callers INVITE. This will also fail if canreinvite is enabled when
|
||||
; the device is actually behind NAT.
|
||||
|
||||
;canreinvite=nonat ; An additional option is to allow media path redirection
|
||||
; (reinvite) but only when the peer where the media is being
|
||||
; sent is known to not be behind a NAT (as the RTP core can
|
||||
; determine it based on the apparent IP address the media
|
||||
; arrives from).
|
||||
; (reinvite) but only when the peer where the media is being
|
||||
; sent is known to not be behind a NAT (as the RTP core can
|
||||
; determine it based on the apparent IP address the media
|
||||
; arrives from).
|
||||
|
||||
;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
|
||||
; instead of INVITE. This can be combined with 'nonat', as
|
||||
; 'canreinvite=update,nonat'. It implies 'yes'.
|
||||
; instead of INVITE. This can be combined with 'nonat', as
|
||||
; 'canreinvite=update,nonat'. It implies 'yes'.
|
||||
|
||||
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
|
||||
; number in SDP packets and will only modify the SDP
|
||||
; session if the version number changes. This option will
|
||||
; force asterisk to ignore the SDP session version number
|
||||
; and treat all SDP data as new data. This is required
|
||||
; for devices that send us non standard SDP packets
|
||||
; (observed with Microsoft OCS). By default this option is
|
||||
; off.
|
||||
; number in SDP packets and will only modify the SDP
|
||||
; session if the version number changes. This option will
|
||||
; force asterisk to ignore the SDP session version number
|
||||
; and treat all SDP data as new data. This is required
|
||||
; for devices that send us non standard SDP packets
|
||||
; (observed with Microsoft OCS). By default this option is
|
||||
; off.
|
||||
|
||||
;----------------------------------------- REALTIME SUPPORT ------------------------
|
||||
; For additional information on ARA, the Asterisk Realtime Architecture,
|
||||
|
@ -689,38 +689,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; source code.
|
||||
;
|
||||
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
|
||||
; just like friends added from the config file only on a
|
||||
; as-needed basis? (yes|no)
|
||||
; just like friends added from the config file only on a
|
||||
; as-needed basis? (yes|no)
|
||||
|
||||
;rtsavesysname=yes ; Save systemname in realtime database at registration
|
||||
; Default= no
|
||||
; Default= no
|
||||
|
||||
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
|
||||
; If set to yes, when a SIP UA registers successfully, the ip address,
|
||||
; the origination port, the registration period, and the username of
|
||||
; the UA will be set to database via realtime.
|
||||
; If not present, defaults to 'yes'. Note: realtime peers will
|
||||
; probably not function across reloads in the way that you expect, if
|
||||
; you turn this option off.
|
||||
; If set to yes, when a SIP UA registers successfully, the ip address,
|
||||
; the origination port, the registration period, and the username of
|
||||
; the UA will be set to database via realtime.
|
||||
; If not present, defaults to 'yes'. Note: realtime peers will
|
||||
; probably not function across reloads in the way that you expect, if
|
||||
; you turn this option off.
|
||||
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
|
||||
; as if it had just registered? (yes|no|<seconds>)
|
||||
; If set to yes, when the registration expires, the friend will
|
||||
; vanish from the configuration until requested again. If set
|
||||
; to an integer, friends expire within this number of seconds
|
||||
; instead of the registration interval.
|
||||
; as if it had just registered? (yes|no|<seconds>)
|
||||
; If set to yes, when the registration expires, the friend will
|
||||
; vanish from the configuration until requested again. If set
|
||||
; to an integer, friends expire within this number of seconds
|
||||
; instead of the registration interval.
|
||||
|
||||
;ignoreregexpire=yes ; Enabling this setting has two functions:
|
||||
;
|
||||
; For non-realtime peers, when their registration expires, the
|
||||
; information will _not_ be removed from memory or the Asterisk database
|
||||
; if you attempt to place a call to the peer, the existing information
|
||||
; will be used in spite of it having expired
|
||||
;
|
||||
; For realtime peers, when the peer is retrieved from realtime storage,
|
||||
; the registration information will be used regardless of whether
|
||||
; it has expired or not; if it expires while the realtime peer
|
||||
; is still in memory (due to caching or other reasons), the
|
||||
; information will not be removed from realtime storage
|
||||
;
|
||||
; For non-realtime peers, when their registration expires, the
|
||||
; information will _not_ be removed from memory or the Asterisk database
|
||||
; if you attempt to place a call to the peer, the existing information
|
||||
; will be used in spite of it having expired
|
||||
;
|
||||
; For realtime peers, when the peer is retrieved from realtime storage,
|
||||
; the registration information will be used regardless of whether
|
||||
; it has expired or not; if it expires while the realtime peer
|
||||
; is still in memory (due to caching or other reasons), the
|
||||
; information will not be removed from realtime storage
|
||||
|
||||
;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
|
||||
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
|
||||
|
@ -744,45 +744,45 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; allowexternaldomains=no
|
||||
|
||||
;domain=mydomain.tld,mydomain-incoming
|
||||
; Add domain and configure incoming context
|
||||
; for external calls to this domain
|
||||
; Add domain and configure incoming context
|
||||
; for external calls to this domain
|
||||
;domain=1.2.3.4 ; Add IP address as local domain
|
||||
; You can have several "domain" settings
|
||||
; You can have several "domain" settings
|
||||
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
|
||||
; Default is yes
|
||||
; Default is yes
|
||||
;autodomain=yes ; Turn this on to have Asterisk add local host
|
||||
; name and local IP to domain list.
|
||||
; name and local IP to domain list.
|
||||
|
||||
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
|
||||
; non-peers, use your primary domain "identity"
|
||||
; for From: headers instead of just your IP
|
||||
; address. This is to be polite and
|
||||
; it may be a mandatory requirement for some
|
||||
; destinations which do not have a prior
|
||||
; account relationship with your server.
|
||||
; non-peers, use your primary domain "identity"
|
||||
; for From: headers instead of just your IP
|
||||
; address. This is to be polite and
|
||||
; it may be a mandatory requirement for some
|
||||
; destinations which do not have a prior
|
||||
; account relationship with your server.
|
||||
|
||||
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
||||
; SIP channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The SIP channel can accept jitter,
|
||||
; thus a jitterbuffer on the receive SIP side will be used only
|
||||
; if it is forced and enabled.
|
||||
; SIP channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The SIP channel can accept jitter,
|
||||
; thus a jitterbuffer on the receive SIP side will be used only
|
||||
; if it is forced and enabled.
|
||||
|
||||
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
|
||||
; channel. Defaults to "no".
|
||||
; channel. Defaults to "no".
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmaxsize) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmaxsize) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
;-----------------------------------------------------------------------------------
|
||||
|
@ -919,7 +919,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;busylevel=2 ; Signal busy at 2 or more calls
|
||||
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
|
||||
;port=80 ; The port number we want to connect to on the remote side
|
||||
; Also used as "defaultport" in combination with "defaultip" settings
|
||||
; Also used as "defaultport" in combination with "defaultip" settings
|
||||
|
||||
;--- sample definition for a provider
|
||||
;[provider1]
|
||||
|
@ -940,30 +940,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; the templates uncommented as they will not harm:
|
||||
|
||||
[basic-options](!) ; a template
|
||||
dtmfmode=rfc2833
|
||||
context=from-office
|
||||
type=friend
|
||||
dtmfmode=rfc2833
|
||||
context=from-office
|
||||
type=friend
|
||||
|
||||
[natted-phone](!,basic-options) ; another template inheriting basic-options
|
||||
nat=yes
|
||||
canreinvite=no
|
||||
host=dynamic
|
||||
nat=yes
|
||||
canreinvite=no
|
||||
host=dynamic
|
||||
|
||||
[public-phone](!,basic-options) ; another template inheriting basic-options
|
||||
nat=no
|
||||
canreinvite=yes
|
||||
nat=no
|
||||
canreinvite=yes
|
||||
|
||||
[my-codecs](!) ; a template for my preferred codecs
|
||||
disallow=all
|
||||
allow=ilbc
|
||||
allow=g729
|
||||
allow=gsm
|
||||
allow=g723
|
||||
allow=ulaw
|
||||
disallow=all
|
||||
allow=ilbc
|
||||
allow=g729
|
||||
allow=gsm
|
||||
allow=g723
|
||||
allow=ulaw
|
||||
|
||||
[ulaw-phone](!) ; and another one for ulaw-only
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
|
||||
; and finally instantiate a few phones
|
||||
;
|
||||
|
@ -982,31 +982,31 @@ allow=ulaw
|
|||
;type=friend
|
||||
;context=from-sip ; Where to start in the dialplan when this phone calls
|
||||
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
|
||||
; on incoming calls to Asterisk
|
||||
; on incoming calls to Asterisk
|
||||
;host=192.168.0.23 ; we have a static but private IP address
|
||||
; No registration allowed
|
||||
; No registration allowed
|
||||
;nat=no ; there is not NAT between phone and Asterisk
|
||||
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
|
||||
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
|
||||
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
|
||||
; from the phone to asterisk (deprecated)
|
||||
; 1 for the explicit peer, 1 for the explicit user,
|
||||
; remember that a friend equals 1 peer and 1 user in
|
||||
; memory
|
||||
; There is no combined call counter for a "friend"
|
||||
; so there's currently no way in sip.conf to limit
|
||||
; to one inbound or outbound call per phone. Use
|
||||
; the group counters in the dial plan for that.
|
||||
;
|
||||
; from the phone to asterisk (deprecated)
|
||||
; 1 for the explicit peer, 1 for the explicit user,
|
||||
; remember that a friend equals 1 peer and 1 user in
|
||||
; memory
|
||||
; There is no combined call counter for a "friend"
|
||||
; so there's currently no way in sip.conf to limit
|
||||
; to one inbound or outbound call per phone. Use
|
||||
; the group counters in the dial plan for that.
|
||||
;
|
||||
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
|
||||
;disallow=all ; need to disallow=all before we can use allow=
|
||||
;allow=ulaw ; Note: In user sections the order of codecs
|
||||
; listed with allow= does NOT matter!
|
||||
; listed with allow= does NOT matter!
|
||||
;allow=alaw
|
||||
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
|
||||
;allow=g729 ; Pass-thru only unless g729 license obtained
|
||||
;callingpres=allowed_passed_screen ; Set caller ID presentation
|
||||
; See README.callingpres for more information
|
||||
; See README.callingpres for more information
|
||||
|
||||
;[xlite1]
|
||||
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
|
||||
|
@ -1035,10 +1035,10 @@ allow=ulaw
|
|||
;defaultip=192.168.0.59 ; IP used until peer registers
|
||||
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
|
||||
;subscribemwi=yes ; Only send notifications if this phone
|
||||
; subscribes for mailbox notification
|
||||
; subscribes for mailbox notification
|
||||
;vmexten=voicemail ; dialplan extension to reach mailbox
|
||||
; sets the Message-Account in the MWI notify message
|
||||
; defaults to global vmexten which defaults to "asterisk"
|
||||
; sets the Message-Account in the MWI notify message
|
||||
; defaults to global vmexten which defaults to "asterisk"
|
||||
;disallow=all
|
||||
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
|
||||
|
||||
|
@ -1051,7 +1051,7 @@ allow=ulaw
|
|||
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
|
||||
;defaultuser=polly ; Username to use in INVITE until peer registers
|
||||
;defaultip=192.168.40.123
|
||||
; Normally you do NOT need to set this parameter
|
||||
; Normally you do NOT need to set this parameter
|
||||
;disallow=all
|
||||
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
|
||||
;progressinband=no ; Polycom phones don't work properly with "never"
|
||||
|
@ -1062,16 +1062,16 @@ allow=ulaw
|
|||
;secret=blah
|
||||
;host=dynamic
|
||||
;insecure=port ; Allow matching of peer by IP address without
|
||||
; matching port number
|
||||
; matching port number
|
||||
;insecure=invite ; Do not require authentication of incoming INVITEs
|
||||
;insecure=port,invite ; (both)
|
||||
;qualify=1000 ; Consider it down if it's 1 second to reply
|
||||
; Helps with NAT session
|
||||
; qualify=yes uses default value
|
||||
; Helps with NAT session
|
||||
; qualify=yes uses default value
|
||||
;qualifyfreq=60 ; Qualification: How often to check for the
|
||||
; host to be up in seconds
|
||||
; Set to low value if you use low timeout for
|
||||
; NAT of UDP sessions
|
||||
; host to be up in seconds
|
||||
; Set to low value if you use low timeout for
|
||||
; NAT of UDP sessions
|
||||
;
|
||||
; Call group and Pickup group should be in the range from 0 to 63
|
||||
;
|
||||
|
@ -1086,30 +1086,30 @@ allow=ulaw
|
|||
;secret=blah
|
||||
;qualify=200 ; Qualify peer is no more than 200ms away
|
||||
;nat=yes ; This phone may be natted
|
||||
; Send SIP and RTP to the IP address that packet is
|
||||
; received from instead of trusting SIP headers
|
||||
; Send SIP and RTP to the IP address that packet is
|
||||
; received from instead of trusting SIP headers
|
||||
;host=dynamic ; This device registers with us
|
||||
;canreinvite=no ; Asterisk by default tries to redirect the
|
||||
; RTP media stream (audio) to go directly from
|
||||
; the caller to the callee. Some devices do not
|
||||
; support this (especially if one of them is
|
||||
; behind a NAT).
|
||||
; RTP media stream (audio) to go directly from
|
||||
; the caller to the callee. Some devices do not
|
||||
; support this (especially if one of them is
|
||||
; behind a NAT).
|
||||
;defaultip=192.168.0.4 ; IP address to use until registration
|
||||
;defaultuser=goran ; Username to use when calling this device before registration
|
||||
; Normally you do NOT need to set this parameter
|
||||
; Normally you do NOT need to set this parameter
|
||||
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
|
||||
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
|
||||
; cause the given audio file to
|
||||
; be played upon completion of
|
||||
; an attended transfer.
|
||||
; cause the given audio file to
|
||||
; be played upon completion of
|
||||
; an attended transfer.
|
||||
|
||||
;[pre14-asterisk]
|
||||
;type=friend
|
||||
;secret=digium
|
||||
;host=dynamic
|
||||
;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
|
||||
; You must have this turned on or DTMF reception will work improperly.
|
||||
; You must have this turned on or DTMF reception will work improperly.
|
||||
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
|
||||
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
|
||||
; external IP address of the remote device. If port forwarding is done at the client side
|
||||
; then UDPTL will flow to the remote device.
|
||||
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
|
||||
; external IP address of the remote device. If port forwarding is done at the client side
|
||||
; then UDPTL will flow to the remote device.
|
||||
|
|
|
@ -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]
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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".
|
||||
;-----------------------------------------------------------------------------------
|
||||
|
|
|
@ -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
|
||||
|
|
Reference in New Issue