asterisk users - Apr 2018 - withheld caller id

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
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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On 04/10/2018 08:02 AM, Atux Atux wrote:
> so any ideas, please?
>
> On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com 
> <mailto:atuxnull at gmail.com>> wrote:
>
>     after adding the ww:
>

See of the D option of dial will do it:

D([called][:calling[:progress]]): Send the specified DTMF strings *after*
 ??? the called party has answered, but before the call gets bridged. The
 ??? <called> DTMF string is sent to the called party, and the? 
<calling> DTMF
 ??? string is sent to the calling party. Both arguments? can be used 
alone.? If
 ??? <progress> is specified, its DTMF is sent immediately after receiving
a
 ??? PROGRESS message.


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Thanks for the reply. So how do i alter my config to call with prefix 9+the
code to block caller id(#31#)+ the number?
now is

exten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT)
exten => _9X.,n,Hangup(${HANGUPCAUSE})



On Wed, Apr 11, 2018 at 11:33 AM, Doug Lytle <support at drdos.info>
wrote:
> On 04/10/2018 08:02 AM, Atux Atux wrote:
>
> so any ideas, please?
>
> On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com>
wrote:
>
>> after adding the ww:
>>
>
>
> See of the D option of dial will do it:
>
> D([called][:calling[:progress]]): Send the specified DTMF strings *after*
>     the called party has answered, but before the call gets bridged. The
>     <called> DTMF string is sent to the called party, and the 
<calling>
> DTMF
>     string is sent to the calling party. Both arguments  can be used
> alone.  If
>     <progress> is specified, its DTMF is sent immediately after
receiving a
>     PROGRESS message.
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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