[asterisk-users] withheld caller id

[Atux Atux]

after adding the ww:
root at Pbx: /etc/asterisk $ asterisk -rvvv
Asterisk 11.25.3, Copyright (C) 1999 - 2013 D  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5                    -- Executing
[9211123456 at AllCalls:1] Goto("SIP/500-00000003",
"DefaultPlan,9211123456,1") in new stack                               --
Goto (DefaultPlan,92105727105,1)
    -- Executing [9211123456 at DefaultPlan:1] Dial("SIP/500-00000003",
"Dongle/dongle800/#31#ww211123456,120,KT") in new stack         [2018-04-10
13:23:46] WARNING[1327][C-00000003]: channel.c:79 parse_dial_string:
Invalid destination '#31#ww211123456' in chan_dongle, only 0123456789*#+ABC
allowed               [2018-04-10 13:23:46] WARNING[1327][C-00000003]:
app_dial.c:2455 dial_exec_full: Unable to create channel of type 'Dongle'
(cause 88 - Incompatible destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [9211123456 at DefaultPlan:2] Hangup("SIP/500-00000003",
"88") in new stack  == Spawn extension (DefaultPlan, 9211123456, 2) exited
non-zero on 'SIP/500-00000003'
Pbx*CLI>

On Tue, Apr 10, 2018 at 1:30 PM, Doug Lytle <support at drdos.info> wrote:

> >>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)
>
> My suggestion would be to add a pause or two before dialing the phone
> number
>
> exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT)
>
> D(digits): After the called party answers, send digits as a DTMF stream,
> then connect the call to the originating channel (you can also use 'w' to
> produce .5 second pauses). You can also provide digits after a colon - all
> digits before the colon are sent to the called channel, all digits after
> the colon are sent to the calling channel (all digits are sent to the
> called channel if there is no colon present).
>
> Doug
>
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