[asterisk-users] withheld caller id

[Atux Atux]

so any ideas, please?

On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote:

> 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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>
>
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