Ringvolume=3 ring volume : 0,1,2,3, can be overrided by Dial(), default = 2 Default = 8Ĭountry=us country (ccTLD) for dial tone frequency. Timeformat=0 0 = 0:00am 1 (default) = 0h00, 2 = 0:00Ĭontrast=8 define the contrast of the LCD. maintext2="(main page)" default = the public IP of the phone, 24 characters maxĭateformat=1 0 = month/day, 1 (default) = day/month maintext1="a custom text" default = the name of the device, 24 characters max none=don't add (default), ask=prompt user, line=use the line number
Only valid in context specified previously. 12 characters maxĮxtension=line Add an extension into the dialplan. Titledefault=TimeZone Americas/Caracas default = "TimeZone (your time zone)". Status_method=0 If you don't see status text, try 1, default = 0 Rtp_method=3 If you don't have sound, you can try 1, 2 or 3, default = 0 Rtp_port=10000 RTP port used by the phone, default = 10000. Linelabel="59202" Softkey label for the next line=> entry, 9 char max. Mailbox=59200 Specify the mailbox number. Maintext0="Cesar en Asterisk" default = "Welcome", 24 characters maxĬontext=admin context, default="default"
#Freepbx install unistim asterisk mac
Extensions configuration-ĭevice=0018b03394da mac address of the phone jblog = no Enables jitterbuffer frame logging. variable size, actually the new jb of IAX2).
(with size always equals to jbmaxsize) and "adaptive" (with
.png "Freepbx install unistim asterisk")
Two implementations are currently available - "fixed" jbimpl = fixed Jitterbuffer implementation, used on the receiving side of a SIP big jumps in/broken timestamps, usually sent from exotic devices Useful to improve the quality of the voice, with jbresyncthreshold = 1000 Jump in the frame timestamps over which the jitterbuffer is jbmaxsize = 200 Max length of the jitterbuffer in milliseconds. jbforce = no Forces the use of a jitterbuffer on the receive side of a SIP thus a jitterbuffer on the receive SIP side will be used only be used only if the sending side can create and the receiving jbenable = yes Enables the use of a jitterbuffer on the receiving side of a autoprovisioning=no Allow undeclared phones to register an extension. public_ip= if asterisk is behind a nat, specify your public IP cos_audio=5 Sets 802.1p priority for RTP audio packets. cos=3 Sets 802.1p priority for signaling packets. tos_audio=ef Sets TOS for RTP audio packets. See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters. This is a working nf config file used in Asterisk 1.6.1 and connected to Nortel IP Phones Model i2002 and i2004. Tar xvjf chan_unistim-0.9.2.tar.bz2 & cd chan_unistim-0.9.2 Install:Ĭhan_unistim is included in the official Asterisk source tree It provides UNISTIM server services that you can use to drive Nortel i2002, i2004, i2007 and i2050 phones.Īvailable on all version of Asterisk.
#Freepbx install unistim asterisk driver
This is a channel driver for the UNISTIM (Unified Networks IP Stimulus) protocol.
0 kommentar(er)
