c369a4416b
* Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls. * Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone * Other described in CHANGES file Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E. (closes issue ASTERISK-16890) Review: https://reviewboard.asterisk.org/r/1243/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
86 lines
5.4 KiB
Text
86 lines
5.4 KiB
Text
;
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; chan_unistim configuration file.
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;
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[general]
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port=5000 ; UDP port
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;
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos=cs3 ; Sets TOS for signaling packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;cos=3 ; Sets 802.1p priority for signaling packets.
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;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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;
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;debug=yes ; Enable debug (default no)
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;keepalive=120 ; in seconds, default = 120
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;public_ip= ; if asterisk is behind a nat, specify your public IP
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;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important
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; informations. no (default), yes, tn.
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;mohsuggest=default
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; SIP channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The SIP channel can accept jitter,
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; thus a jitterbuffer on the receive SIP side will be used only
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; if it is forced and enabled.
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; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
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; channel. Defaults to "no".
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmaxsize) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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;[black] ; name of the device
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;device=000ae4012345 ; mac address of the phone
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;rtp_port=10000 ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
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;rtp_method=0 ; If you don't have sound, you can try 1, 2 or 3, default = 0
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; value 3 works on newer i2004, 1120E and 1140E
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;status_method=0 ; If you don't see status text, try 1, default = 0
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; value 1 works on 1120E and 1140E
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;titledefault=Asterisk ; default = "TimeZone (your time zone)". 12 characters max
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;height=3 ; default = 3, the number of display lines the device can show
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; For example on a Nortel I2001 or I2002, set this to 1
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;maintext0="you can insert" ; default = "Welcome", 24 characters max
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;maintext1="a custom text" ; default = the name of the device, 24 characters max
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;maintext2="(main page)" ; default = the public IP of the phone, 24 characters max
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;dateformat=1 ; 0 = month/day, 1 (default) = day/month
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;timeformat=1 ; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00
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;contrast=8 ; define the contrast of the LCD. From 0 to 15. Default = 8
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;country=us ; country (ccTLD) for dial tone frequency. See README, default = us
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;ringvolume=2 ; ring volume : 0,1,2,3, can be overrided by Dial(), default = 2
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;ringstyle=3 ; ring style : 0 to 7, can be overrided by Dial(), default = 3
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;cwvolume=2 ; ring volume : 0,1,2,3, default = 0
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;cwstyle=3 ; ring style : 0 to 7, default = 2
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;sharpdial=1 ; dial number by pressing #, default = 0
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;callhistory=1 ; 0 = disable, 1 = enable call history, default = 1
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;callerid="Customer Support" <555-234-5678>
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;context=default ; context, default="default"
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;mailbox=1234 ; Specify the mailbox number. Used by Message Waiting Indication
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;linelabel="Support" ; Softkey label for the next line=> entry, 9 char max.
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;extension=none ; Add an extension into the dialplan. Only valid in context specified previously.
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; none=don't add (default), ask=prompt user, line=use the line number
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;line => 100 ; Any number of lines can be defined in any order with bookmarks
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;line => 200 ; After line defined it placed in next available slot
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;bookmark=Hans C.@123 ; Use a softkey to dial 123. Name : 9 char max
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;bookmark=Mailbox@011@54 ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63)
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;bookmark=Test@*@USTM/violet ; Display an icon if violet is connected (dynamic), only for unistim device
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;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed
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;[violet]
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;device=006038abcdef
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;line => 102
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