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Author Topic: i2004 Paging / Intercom with Asterisk issue  (Read 124 times)

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Offline cityitguy

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i2004 Paging / Intercom with Asterisk issue
« on: October 14, 2019, 04:00:58 PM »
I have a non-profit whose Nortel phone system died. I wanted to replace it with an emetrotel box, but they have no money to work with so I got their i2004 phones up and running using an old computer and the FreePBX version of asterisk. I have all of the features they need working with one exception, I cannot the the "Page/Intercom" feature to work with the phones. I setup the page group and when you dial the number for the page group, all of the phones ring once and then silence instead of the usual "beep" and going into speaker mode so an announcement can be made. I know the e-metrotel folks got this feature to work with these phones. I know the unistim protocol was reverse engineered for asterisk, and not all features were supported.  Does anyone have any ideas how to get this working? If I can figure out how to get this going, I think the Asterisk solution will be sufficient for them.  Here is a sample of my one of my extensions in the unistim.conf file.  Any assistance is greatly appreciated.


; chan_unistim configuration file.
;

[general]
port=5000 ; UDP port
;
;
;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".
;-----------------------------------------------------------------------------------

[721]                     ; name of the device
device=000AE4705d60         ; mac address of the phone
;tn=721
maintext0="Extension 721"  ; default = "Welcome", 24 characters max
callerid="Extension 721" <721>
context=from-internal           ; context, default="default"
mailbox=721                ; Specify the mailbox number. Used by Message Waiting Indication
linelabel="My Line"         ; Softkey label for the next line=> entry, 9 char max.
rtp_port=10000              ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
rtp_method=3                ; If you don't have sound, you can try 1, 2 or 3, default = 0
status_method=0             ; If you don't see status text, try 1, default = 0
titledefault=Manwarings ; default = "TimeZone (your time zone)". 12 characters max
extension=line              ; 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
maintext1="Brian's Office"   ; 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 = month/day, 1 (default) = day/month
timeformat=0                ; 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
ringvolume=3                ; 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
;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
line => 721                 ; Only one line by device is currently supported.
                             ; Beware ! only bookmark and softkey entries are allowed after line=>
bookmark=1@All Page@800@51   ; Display a pager icon and dial 54321 when softkey 4 is pressed
bookmark=Dorothy@5216025        ; Use a softkey to dial 123. Name : 9 char max
bookmark=Veronica@2016168     ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63)
bookmark=Brian@3036676 ; Display an icon if violet is connected (dynamic), only for unistim device
bookmark=Ron@5343808 ; Display an icon if violet is connected (dynamic), only for unistim device

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