2 ; chan_unistim configuration file.
8 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
9 ;tos=cs3 ; Sets TOS for signaling packets.
10 ;tos_audio=ef ; Sets TOS for RTP audio packets.
11 ;cos=3 ; Sets 802.1p priority for signaling packets.
12 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
14 ;keepalive=120 ; in seconds, default = 120
15 ;public_ip= ; if asterisk is behind a nat, specify your public IP
16 ;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important
17 ; informations. no (default), yes, tn.
18 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
19 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
20 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
21 ; be used only if the sending side can create and the receiving
22 ; side can not accept jitter. The SIP channel can accept jitter,
23 ; thus a jitterbuffer on the receive SIP side will be used only
24 ; if it is forced and enabled.
26 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
27 ; channel. Defaults to "no".
29 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
31 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
32 ; resynchronized. Useful to improve the quality of the voice, with
33 ; big jumps in/broken timestamps, usually sent from exotic devices
34 ; and programs. Defaults to 1000.
36 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
37 ; channel. Two implementations are currently available - "fixed"
38 ; (with size always equals to jbmaxsize) and "adaptive" (with
39 ; variable size, actually the new jb of IAX2). Defaults to fixed.
41 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
42 ;-----------------------------------------------------------------------------------
45 ;[black] ; name of the device
46 ;device=000ae4012345 ; mac address of the phone
47 ;rtp_port=10000 ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
48 ;rtp_method=0 ; If you don't have sound, you can try 1, 2 or 3, default = 0
49 ;status_method=0 ; If you don't see status text, try 1, default = 0
50 ;titledefault=Asterisk ; default = "TimeZone (your time zone)". 12 characters max
51 ;maintext0="you can insert" ; default = "Welcome", 24 characters max
52 ;maintext1="a custom text" ; default = the name of the device, 24 characters max
53 ;maintext2="(main page)" ; default = the public IP of the phone, 24 characters max
54 ;dateformat=1 ; 0 = month/day, 1 (default) = day/month
55 ;timeformat=1 ; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00
56 ;contrast=8 ; define the contrast of the LCD. From 0 to 15. Default = 8
57 ;country=us ; country (ccTLD) for dial tone frequency. See README, default = us
58 ;ringvolume=2 ; ring volume : 0,1,2,3, can be overrided by Dial(), default = 2
59 ;ringstyle=3 ; ring style : 0 to 7, can be overrided by Dial(), default = 3
60 ;callhistory=1 ; 0 = disable, 1 = enable call history, default = 1
61 ;callerid="Customer Support" <555-234-5678>
62 ;context=default ; context, default="default"
63 ;mailbox=1234 ; Specify the mailbox number. Used by Message Waiting Indication
64 ;linelabel="Support" ; Softkey label for the next line=> entry, 9 char max.
65 ;extension=none ; Add an extension into the dialplan. Only valid in context specified previously.
66 ; none=don't add (default), ask=prompt user, line=use the line number
67 ;line => 100 ; Only one line by device is currently supported.
68 ; Beware ! only bookmark and softkey entries are allowed after line=>
69 ;bookmark=Hans C.@123 ; Use a softkey to dial 123. Name : 9 char max
70 ;bookmark=Mailbox@011@54 ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63)
71 ;bookmark=Test@*@USTM/violet ; Display an icon if violet is connected (dynamic), only for unistim device
72 ;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed