diff options
author | Matthew Jordan <mjordan@digium.com> | 2014-07-17 21:17:28 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2014-07-17 21:17:28 +0000 |
commit | fc0fecb4768d696db3324bcf6dd03325bb4cd513 (patch) | |
tree | 12615f96e88382b2824d4901f6949571e41ea2e4 /configs/samples/unistim.conf.sample | |
parent | 1ce23d4534994fdd8bfb8ad3b9ca1884194097be (diff) |
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
Diffstat (limited to 'configs/samples/unistim.conf.sample')
-rw-r--r-- | configs/samples/unistim.conf.sample | 88 |
1 files changed, 88 insertions, 0 deletions
diff --git a/configs/samples/unistim.conf.sample b/configs/samples/unistim.conf.sample new file mode 100644 index 000000000..c33426b0c --- /dev/null +++ b/configs/samples/unistim.conf.sample @@ -0,0 +1,88 @@ +; +; 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 |