NO JOY Chan_Unistim sent to an invalid context on dial out

[chwebb1]

I'm trying to figure out how to get a Nortel i2007 to be able to dial out. So far, I am able to make and receive calls from a local extension, but when I try to dial out from the i2007, I get the "You have dialed an invalid extension blah blah blah" playing on the phone, and this in the log:

[2014-03-07 00:00:23] VERBOSE[23076][C-0000014b] pbx.c: -- Channel 'USTM/0020@black-0x7fca3c12e2c0' sent to invalid extension: context,exten,priority=default,18004212110,1
[2014-03-07 00:00:23] VERBOSE[23076][C-0000014b] pbx_lua.c: -- Executing [i@default:1] playback("USTM/0020@black-0x7fca3c12e2c0", "invalid")
[2014-03-07 00:00:23] VERBOSE[23076][C-0000014b] file.c: -- <USTM/0020@black-0x7fca3c12e2c0> Playing 'invalid.gsm' (language 'en')
[2014-03-07 00:00:28] VERBOSE[23076][C-0000014b] pbx_lua.c: -- Executing [i@default:1] background("USTM/0020@black-0x7fca3c12e2c0", "demo-instruct")
[2014-03-07 00:00:28] VERBOSE[23076][C-0000014b] file.c: -- <USTM/0020@black-0x7fca3c12e2c0> Playing 'demo-instruct.gsm' (language 'en')
[2014-03-07 00:00:32] VERBOSE[23076][C-0000014b] pbx.c: == Spawn extension (default, i, 1) exited non-zero on 'USTM/0020@black-0x7fca3c12e2c0'
[2014-03-07 00:00:32] VERBOSE[23076][C-0000014b] pbx.c: -- Executing [h@default:1] Macro("USTM/0020@black-0x7fca3c12e2c0", "hangupcall,") in new stack
[2014-03-07 00:00:32] VERBOSE[23076][C-0000014b] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("USTM/0020@black-0x7fca3c12e2c0", "1?theend") in new stack
[2014-03-07 00:00:32] VERBOSE[23076][C-0000014b] pbx.c: -- Goto (macro-hangupcall,s,3)

Dialing this number on a SIP phone works as expected (the call connects). I have attached the relevant config from my unistim.conf file here:

[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=00159be3b475        ; mac address of the phone
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
                            ; 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 = 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=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
;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="USTM Phone" <0020>
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=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
line => 0020                ; 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

Does anyone have any idea what might be going on?
Thanks!

 

[chwebb1]

In case anyone else has this issue, I created a new context using Custom Contexts that allowed access to all outbound trunks and used that as the context in Chan-UNISTIM. That fixed this issue for me.