; ; chan_unistim configuration file. ; [general] ;port=5002 ; UDP port ; ; See qos.tex or Quality of Service section of asterisk.pdf 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. ; ;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. ;------------------------------ 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 ;status_method=0 ; If you don't see status text, try 1, default = 0 ;titledefault=Asterisk ; default = "TimeZone (your time zone)". 12 characters max ;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=1 ; 0 = month/day, 1 (default) = 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 ;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 ;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 ; Only one line by device is currently supported. ; 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 ;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed ;[violet] ;device=006038abcdef ;line => 102