Re: [asterisk-users] withheld caller id

[Doug Lytle]

>>> 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?

That's something I'll leave for you to investigate.  As many have said, "Google 
is your friend"

Doug

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Re: [asterisk-users] withheld caller id

[Atux Atux]

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  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  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
>  DTMF string is sent to the called party, and the  
> DTMF
> string is sent to the calling party. Both arguments  can be used
> alone.  If
>  is specified, its DTMF is sent immediately after receiving a
> PROGRESS message.
>
>
>
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> org/
>
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Re: [asterisk-users] withheld caller id

[Doug Lytle]

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 > 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
     DTMF string is sent to the called party, and the  
 DTMF
    string is sent to the calling party. Both arguments  can be used 
alone.  If

     is specified, its DTMF is sent immediately after receiving a
    PROGRESS message.


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Re: [asterisk-users] withheld caller id

[Atux Atux]

so any ideas, please?

On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux  wrote:

> after adding the ww:
> root@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@AllCalls:1] Goto("SIP/500-0003",
> "DefaultPlan,9211123456,1") in new stack   --
> Goto (DefaultPlan,92105727105,1)
> -- Executing [9211123456@DefaultPlan:1] Dial("SIP/500-0003",
> "Dongle/dongle800/#31#ww211123456,120,KT") in new stack
> [2018-04-10 13:23:46] WARNING[1327][C-0003]: 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-0003]: 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@DefaultPlan:2] Hangup("SIP/500-0003",
> "88") in new stack  == Spawn extension (DefaultPlan, 9211123456, 2) exited
> non-zero on 'SIP/500-0003'
> Pbx*CLI>
>
> On Tue, Apr 10, 2018 at 1:30 PM, Doug Lytle  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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Re: [asterisk-users] withheld caller id

[Atux Atux]

after adding the ww:
root@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@AllCalls:1] Goto("SIP/500-0003",
"DefaultPlan,9211123456,1") in new stack   --
Goto (DefaultPlan,92105727105,1)
-- Executing [9211123456@DefaultPlan:1] Dial("SIP/500-0003",
"Dongle/dongle800/#31#ww211123456,120,KT") in new stack [2018-04-10
13:23:46] WARNING[1327][C-0003]: 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-0003]:
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@DefaultPlan:2] Hangup("SIP/500-0003",
"88") in new stack  == Spawn extension (DefaultPlan, 9211123456, 2) exited
non-zero on 'SIP/500-0003'
Pbx*CLI>

On Tue, Apr 10, 2018 at 1:30 PM, Doug Lytle  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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Re: [asterisk-users] withheld caller id

[Doug Lytle]

>>> My suggestion would be to add a pause or two before dialing the phone number

Looks like using w for a pause is no longer supported.

Doug

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Re: [asterisk-users] withheld caller id

[Doug Lytle]

>>> > 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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Re: [asterisk-users] withheld caller id

[ka]

On 2018-04-10 10:19, Atux Atux wrote:


exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)
exten => _9X.,n,Hangup(${HANGUPCAUSE})



What am i doing wrong in asterisk?



unless i'm missing something your config looks OK.  Do you have any logs 
/ debugs of what number is actually being dialed?


--
Regards
Adam


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Re: [asterisk-users] withheld caller id

[Atux Atux]

thanks a lot for the reply.
i thought of that and i did try to send

*exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)exten =>
_9X.,n,Hangup(${HANGUPCAUSE})*

but the provider replies back that it is a wrong number. Then i inserted
the sim to an ordinary mobile phone and dialed #31# and the number, then
the call progressed fine and it restricted the number.
What am i doing wrong in asterisk?

On Tue, Apr 10, 2018 at 11:43 AM,  wrote:

> On 2018-04-10 08:46, Atux Atux wrote:
>
>> 9+#31#+destination_number. Unfortunately, zoiper did stop on 9#31# and
>> it dialled one of my recent numbers. The same result happened with
>>
>
> haven't used zoiper at all, so can't comment on its features of parsing
> numbers.  I'd recommend 'hiding' this function or making it transparent to
> the end user by using something like this:
>
> exten => _06[237]0NXX!,100,Dial(SIP/${OUTGOING_PROVIDER}/*31#0036
> ${EXTEN:2},55)
>
> where ${OUTGOING_PROVIDER} is set by a macro previously, and *31# i
> believe is the caller ID set visible.  I have used it with #31# as well but
> the customer requirements have changed and they now want the number to be
> visible at all times.
>
> Because the #31# or *31# is transparent to the end user and won't have do
> dial it at all, it doesn't matter if zoiper intercepts digits and parses
> them on its own.
>
> If you want the end user to be able to control when the number is
> shown/hidden, i'd recommend using either a pefix (90 for hiding, 91 for
> showing), or use an SQL backend from where an extensions.conf macro can
> fetch the current settings (maybe even profile people).
>
> --
> Regards
> Adam
>
>
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Re: [asterisk-users] withheld caller id

[ka]

On 2018-04-10 08:46, Atux Atux wrote:

9+#31#+destination_number. Unfortunately, zoiper did stop on 9#31# and
it dialled one of my recent numbers. The same result happened with


haven't used zoiper at all, so can't comment on its features of parsing 
numbers.  I'd recommend 'hiding' this function or making it transparent 
to the end user by using something like this:


exten => 
_06[237]0NXX!,100,Dial(SIP/${OUTGOING_PROVIDER}/*31#0036${EXTEN:2},55)


where ${OUTGOING_PROVIDER} is set by a macro previously, and *31# i 
believe is the caller ID set visible.  I have used it with #31# as well 
but the customer requirements have changed and they now want the number 
to be visible at all times.


Because the #31# or *31# is transparent to the end user and won't have 
do dial it at all, it doesn't matter if zoiper intercepts digits and 
parses them on its own.


If you want the end user to be able to control when the number is 
shown/hidden, i'd recommend using either a pefix (90 for hiding, 91 for 
showing), or use an SQL backend from where an extensions.conf macro can 
fetch the current settings (maybe even profile people).


--
Regards
Adam


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