From fc0fecb4768d696db3324bcf6dd03325bb4cd513 Mon Sep 17 00:00:00 2001 From: Matthew Jordan Date: Thu, 17 Jul 2014 21:17:28 +0000 Subject: configs: Move sample config files into a subdirectory of configs This moves all samples configs from configs/ to configs/samples. This allows for additional sets of sample configuration files to be added in the future. Review: https://reviewboard.asterisk.org/r/3804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/unistim.conf.sample | 88 --------------------------------------------- 1 file changed, 88 deletions(-) delete mode 100644 configs/unistim.conf.sample (limited to 'configs/unistim.conf.sample') diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample deleted file mode 100644 index c33426b0c..000000000 --- a/configs/unistim.conf.sample +++ /dev/null @@ -1,88 +0,0 @@ -; -; 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 -- cgit v1.2.3