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authorMatthew Jordan <mjordan@digium.com>2014-07-17 21:17:28 +0000
committerMatthew Jordan <mjordan@digium.com>2014-07-17 21:17:28 +0000
commitfc0fecb4768d696db3324bcf6dd03325bb4cd513 (patch)
tree12615f96e88382b2824d4901f6949571e41ea2e4 /configs/unistim.conf.sample
parent1ce23d4534994fdd8bfb8ad3b9ca1884194097be (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/unistim.conf.sample')
-rw-r--r--configs/unistim.conf.sample88
1 files changed, 0 insertions, 88 deletions
diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample
deleted file mode 100644
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--- 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