;
; chan_unistim configuration file.
;

[general]
port=5000                    ; UDP port
;
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos=cs3                ; Sets TOS for signaling packets.
;tos_audio=ef           ; Sets TOS for RTP audio packets.
;cos=3                  ; Sets 802.1p priority for signaling packets.
;cos_audio=5            ; Sets 802.1p priority for RTP audio packets.
;
;debug=yes              ; Enable debug (default no)
;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.
;mohsuggest=default
; ----------------------------- 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.

; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
                              ; 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.

; 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.

; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
; ----------------------------------------------------------------------------------


;[black]                     ; name of the device
;device=000ae4012345         ; mac address of the phone
;rtp_port=10000              ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
;rtp_method=0                ; If you don't have sound, you can try 1, 2 or 3, default = 0
                             ; value 3 works on newer i2004, 1120E and 1140E
;status_method=0             ; If you don't see status text, try 1, default = 0
                             ; value 1 works on 1120E and 1140E
;titledefault=Asterisk       ; default = "TimeZone (your time zone)". 12 characters max
;height=3                    ; default = 3, the number of display lines the device can show
                             ; For example on a Nortel I2001 or I2002, set this to 1
;maintext0="you can insert"  ; default = "Welcome", 24 characters max
;maintext1="a custom text"   ; default = the name of the device, 24 characters max
;maintext2="(main page)"     ; default = the public IP of the phone, 24 characters max
;dateformat=0                ; 0 (default) = 31Jan, 1 = Jan31, 2 = month/day, 3 = day/month
;timeformat=1                ; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00
;contrast=8                  ; define the contrast of the LCD. From 0 to 15. Default = 8
;country=us                  ; country (ccTLD) for dial tone frequency. See README, default = us
;language=ru                 ; language used for audio files and onscreen messages translate
;ringvolume=2                ; ring volume : 0,1,2,3, can be overrided by Dial(), default = 2
;ringstyle=3                 ; ring style : 0 to 7, can be overrided by Dial(), default = 3
;cwvolume=2                  ; ring volume : 0,1,2,3, default = 0
;cwstyle=3                   ; ring style : 0 to 7, default = 2
;sharpdial=1                 ; dial number by pressing #, default = 0
;dtmf_duration=0             ; DTMF playback duration (in milliseconds) 0..150 (0 = off (default), 150 = maximum)
;interdigit_timer=4000       ; timer for automatic dial after several digits of number entered (in ms, 0 is off)
;callhistory=1               ; 0 = disable, 1 = enable call history, default = 1
;callerid="Customer Support" <555-234-5678>
;context=default             ; context, default="default"
;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
;line => 100                 ; Any number of lines can be defined in any order with bookmarks
;line => 200                 ; After line defined it placed in next available slot
;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
;bookmark=4@Pager@54321@51   ; Display a pager icon and dial 54321 when softkey 4 is pressed

;[violet]
;device=006038abcdef
;line => 102
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  • asterisk/cf/unistim.conf.txt
  • Последние изменения: 2016/11/12