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 <supp...@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 <atuxn...@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
>  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.
>
>
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] withheld caller id

[Atux Atux]

so any ideas, please?

On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxn...@gmail.com> 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 <supp...@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
>>
>
>
-- 
_
-- 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

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
>
-- 
_
-- 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

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, <k...@mayten.sch.bme.hu> 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
>
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] withheld caller id

[Atux Atux]

Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls
with the following config in extensions.conf

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

By dialing 9 it opens the dongle to make a call.
I would like to restrict my caller id. so when i place a call from this
dongle, it will send on the other end *blocked number* or *withheld*. The
carrier restricts (call by call) the caller ID if i send #31# before the
number (that works if i place the sim in my mobile)*. *i did try to the
following from zoiper
9*+*#31#*+*destination_number. Unfortunately, zoiper did stop on 9#31# and
it dialled one of my recent numbers. The same result happened with other
softphone clients.
I would like to restrict my callerID when placing calls from this dongle,
so i would like your help please.
-- 
_
-- 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

Re: [asterisk-users] Asterisk as gateway

[Atux Atux]

This is a setup of Asterisk as extension to an existing Asterisk PBX. It
has to be that way and not IAX. Simply we need to an extension number with
DIDs to an external PBX which is a helper to our office. This has to be
done for the second PBX as well.


On Thu, Mar 22, 2018 at 2:18 PM, Atux Atux <atuxn...@gmail.com> wrote:

> i would like to ask how to connect 2 systems. I would like to have an
> asterisk where it will have all the connections to the outside world (sip
> trunks) and it will called the gateway. This asterisk will have extension
> numbers of 3XX.
> In the LAN there will be 2 other asterisk boxes (A & B) where A will have
> the extension numbers 4XX and B the 5XX.
> -gateway 3XX has all sip trunks to the outside world
> -A 4XX.
> -B 5XX
>
> I would like to have A to connect to the gateway as extension 308 and
> route all calls incoming/outgoing through the gateway. the same applies to
> B as extension 309.
> i am kinda lost with config and the dialplan. In the gateway i have in the
> sip.conf 2extensions 308 &309. in the gateway's extension.conf i have 5
> DIDs for 308 and another 5 for 309 as follows:
>
> 308's first DID up to 123456784
> exten => 123456780,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
> exten => 123456780,n,Answer()
> exten => 123456780,n,Wait(1)
> exten => 123456780,n,Dial(SIP/308,20)
> exten => 123456780,n,VoiceMail(308@home,u)
> exten => 123456780,n,Busy(3)
>
>
>
>
>
> 309's first DID up to 123456789
> exten => 123456785,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
> exten => 123456785,n,Answer()
> exten => 123456785,n,Wait(1)
> exten => 123456785,n,Dial(SIP/309,20)
> exten => 123456785,n,VoiceMail(309@home,u)
> exten => 123456785,n,Busy(3)
>
>
>
> Some help please?
> John
>
-- 
_
-- 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

Re: [asterisk-users] invite to conference by a call file

[Atux Atux]

that's the problem. it is never the same people

On Thu, Mar 22, 2018 at 3:15 PM, Frank Vanoni <mailingl...@linuxista.com>
wrote:

> If the participants are always the same people, there is no need to
> change the dialplan. Just tells the office secretary "Please, place a
> conference call.". with the "Page" application, she picks up the phone,
> dials a predefined number and all the participants are called at once.
> Easy peasy. :-)
>
>
> On Thu, 2018-03-22 at 14:21 +0200, Atux Atux wrote:
> > All the aforementioned techniques need change everytime on the
> > dialplan. I need the office secretary to edit a file (call file) and
> > place it in a particular folder in their windows PCs. this folder is
> > the outgoing folder of LINUX shared through samba in LAN. i need to
> > make it as easy as possible, please.
> >
> > On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist@linuxista.
> > com> wrote:
> > > Here I'm using the "Page" application to make a conference call "on
> > > the fly".
> > >
> > >
> > >
> > > [office]
> > >
> > > exten => ,1,Dial(SIP/desk2,150)
> > >same => n,Hangup()
> > >
> > > exten => ,1,Dial(SIP/desk3,150)
> > >same => n,Hangup()
> > >
> > > exten => ,1,Dial(SIP/desk4,150)
> > >same => n,Hangup()
> > >
> > > exten => ,1,Dial(SIP/desk5,150)
> > >same => n,Hangup()
> > >
> > > exten => ,1,Dial(SIP/desk6,150)
> > >same => n,Hangup()
> > >
> > > ; Conference call
> > > exten => ,1,Answer
> > > exten => ,n,Page(Local/@office/@office/
> > > @office/@office/@office,d)
> > > same => n,Hangup()
> > >
> > > --
> > > ___
> > > __
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> > > --
> > >
> > > Check out the new Asterisk community forum at: https://community.as
> > > terisk.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
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at: https://community.aste
> > risk.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
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] invite to conference by a call file

[Atux Atux]

All the aforementioned techniques need change everytime on the dialplan. I
need the office secretary to edit a file (call file) and place it in a
particular folder in their windows PCs. this folder is the outgoing folder
of LINUX shared through samba in LAN. i need to make it as easy as
possible, please.

On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni 
wrote:

> Here I'm using the "Page" application to make a conference call "on the
> fly".
>
>
>
> [office]
>
> exten => ,1,Dial(SIP/desk2,150)
>same => n,Hangup()
>
> exten => ,1,Dial(SIP/desk3,150)
>same => n,Hangup()
>
> exten => ,1,Dial(SIP/desk4,150)
>same => n,Hangup()
>
> exten => ,1,Dial(SIP/desk5,150)
>same => n,Hangup()
>
> exten => ,1,Dial(SIP/desk6,150)
>same => n,Hangup()
>
> ; Conference call
> exten => ,1,Answer
> exten => ,n,Page(Local/@office/@office/@office
> /@office/@office,d)
> same => n,Hangup()
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] Asterisk as gateway

[Atux Atux]

i would like to ask how to connect 2 systems. I would like to have an
asterisk where it will have all the connections to the outside world (sip
trunks) and it will called the gateway. This asterisk will have extension
numbers of 3XX.
In the LAN there will be 2 other asterisk boxes (A & B) where A will have
the extension numbers 4XX and B the 5XX.
-gateway 3XX has all sip trunks to the outside world
-A 4XX.
-B 5XX

I would like to have A to connect to the gateway as extension 308 and route
all calls incoming/outgoing through the gateway. the same applies to B as
extension 309.
i am kinda lost with config and the dialplan. In the gateway i have in the
sip.conf 2extensions 308 &309. in the gateway's extension.conf i have 5
DIDs for 308 and another 5 for 309 as follows:

308's first DID up to 123456784
exten => 123456780,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => 123456780,n,Answer()
exten => 123456780,n,Wait(1)
exten => 123456780,n,Dial(SIP/308,20)
exten => 123456780,n,VoiceMail(308@home,u)
exten => 123456780,n,Busy(3)





309's first DID up to 123456789
exten => 123456785,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => 123456785,n,Answer()
exten => 123456785,n,Wait(1)
exten => 123456785,n,Dial(SIP/309,20)
exten => 123456785,n,VoiceMail(309@home,u)
exten => 123456785,n,Busy(3)



Some help please?
John
-- 
_
-- 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

Re: [asterisk-users] invite to conference by a call file

[Atux Atux]

thanks a lot for the reply.
[call-file-test]
Exten => 10,1,Answer
 same => ConfBridge(100)


i assume 100 is the conference room, correct?
where do i write the SIP numbers to invite(internal or external)?
what about the PIN?


On Tue, Mar 20, 2018 at 4:41 PM, Dovid Bender <do...@telecurve.com> wrote:

> Atux,
>
> This should work:
> [call-file-test]
> Exten => 10,1,Answer
>  same => ConfBridge(100)
>
> On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux <atuxn...@gmail.com> wrote:
>
>> Hi. in my system i have a conference room where someone can call it eg
>> 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in
>> through a different number and PIN.  I would like to have a call file and
>> call all participants eg 610-619 at certain time of the day and give them
>> access to the conference.
>> During my try i managed to create a call file where it calls the a SIP
>> phone and it can hear the monkeys (just for test).
>> here is the call file
>> Channel: SIP/601
>> MaxRetries: 2
>> RetryTime: 60
>> WaitTime: 30
>> Context: call-file-test
>> Extension: 10
>>
>>
>>
>> and here is the entry in extensions.conf
>>
>> [call-file-test]
>> exten => 10,1,Answer()
>> exten => 10,n,Wait(1)
>> exten => 10,n,Playback(tt-monkeys)
>> exten => 10,n,Wait(1)
>> exten => 10,n,Hangup()
>>
>>
>> i did not manage to make it call more SIP phones and invite them to the
>> conference
>>
>> Any ideas please?
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] invite to conference by a call file

[Atux Atux]

Hi. in my system i have a conference room where someone can call it eg 698
dial the PIN eg 1234 and enter the room as a user. The admin enters in
through a different number and PIN.  I would like to have a call file and
call all participants eg 610-619 at certain time of the day and give them
access to the conference.
During my try i managed to create a call file where it calls the a SIP
phone and it can hear the monkeys (just for test).
here is the call file
Channel: SIP/601
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: call-file-test
Extension: 10



and here is the entry in extensions.conf

[call-file-test]
exten => 10,1,Answer()
exten => 10,n,Wait(1)
exten => 10,n,Playback(tt-monkeys)
exten => 10,n,Wait(1)
exten => 10,n,Hangup()


i did not manage to make it call more SIP phones and invite them to the
conference

Any ideas please?
-- 
_
-- 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

Re: [asterisk-users] Bank holidays read from file?

[Atux Atux]

Hi all,


i have configured my calendar.conf accordingly to the Asterisk-the
definitive guide 4th edition and even though i can see the calendar, it
always shows me as free.
as you can see in the attached printscreen from my calendar, it has 3
bookings today, but the asterisk calendar shows me as free.  Here is what's
going on

PBX*CLI> calendar show calendar myGoogleCal
Name  : myGoogleCal
Notify channel: SIP/601
Notify context: calendar
Notify extension  :
Notify applicatio :
Notify appdata:
Refresh time  : 15
Timeframe : 180
Autoreminder  : 10
Events
--
PBX*CLI> calendar show calendars
Calendar Type   Status
    --
myGoogleCal  caldav free
PBX*CLI>


-
*calendar.conf*
[myGoogleCal]
type=caldav
url=
https://calendar.google.com/calendar/ical/atuxnull%40gmail.com/private-ad3e9197080e5ef00c6155f150fdf1aa/basic.ics
user=atuxn...@gmail.com
secret=superdoopersecrete
refresh=15
timeframe=180
autoreminder=10
channel=SIP/601
context=calendar


On Thu, Mar 15, 2018 at 5:41 PM, Ludovic Gasc <gml...@gmail.com> wrote:

> An example, you need to change the URL with Google ical URL.
>
> [tests.28]
> type = ical  ;  type of calendar--currently supported: ical,
> caldav, exchange, or ews
> url = http://127.0.0.1:7999/tests/cache/calendars/28.ics   ; URL to
> shared calendar (Zimbra example)
> refresh = 1 ; refresh calendar every n minutes
> timeframe = 1   ; number of minutes of calendar data to pull for
> each refresh period
>
>
>
> --
> Ludovic Gasc (GMLudo)
>
> 2018-03-15 15:40 GMT+01:00 Atux Atux <atuxn...@gmail.com>:
>
>> Hi. thanks a lot for your reply. i will download the newer libical
>> software. Could you elaborate on icalendar with google calendar config and
>> calendar.conf, please?
>>
>>
>> On Thu, Mar 15, 2018 at 3:00 PM, Ludovic Gasc <gml...@gmail.com> wrote:
>>
>>> I never use caldav mode, always icalendar with Google Calendar.
>>>
>>> BTW, you use old versions of libical, Asterisk and Debian, I recommend
>>> you to upgrade or install a new server with Debian Stretch: You will have
>>> Asterisk 13, libical2 and security upgrades.
>>> Asterisk 11 doesn't have security fixes anymore.
>>>
>>> Regards.
>>>
>>> --
>>> Ludovic Gasc (GMLudo)
>>>
>>> 2018-03-15 11:28 GMT+01:00 Atux Atux <atuxn...@gmail.com>:
>>>
>>>> Hi. Thanks for the idea for calendar, it sounds better. i did not
>>>> manage to make it work though. i am running debian 8 32 bit with asterisk
>>>> 11.25.3. I have installed the packages libneon27-dev & libical-dev then in
>>>> /etc/asterisk the file calendar.conf has the following entries:
>>>> [Gcalendar]
>>>>
>>>> type=caldav
>>>> url=https://www.google.com/calendar/dav/atuxn...@gmail.com/events/
>>>> user=atuxn...@gmail.com
>>>> secret=MySuperDooperPasswd
>>>> refresh=15
>>>>
>>>>
>>>>
>>>> then a reload to the system and try to see:
>>>> PBX> calendar show calendars
>>>> Calendar Type Status
>>>>   --
>>>>
>>>>
>>>>
>>>> there is nothing shown over here. Off course i checked if my calendar
>>>> is public but i have not an idea why it is not working.
>>>>
>>>>
>>>>
>>>> On Tue, Mar 13, 2018 at 10:23 PM, Ludovic Gasc <gml...@gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I recommend you to use calendar module of Asterisk:
>>>>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Calendaring
>>>>>
>>>>> We are using in production since two years with several hundred
>>>>> calendars, it works pretty well.
>>>>> However, I strongly recommend you to use the latest stable version of
>>>>> libical3, because they have fixed a lot of bugs, especially with recurring
>>>>> events.
>>>>>
>>>>> And to use Asterisk 15: We had time to time crashes with Asterisk 13
>>>>> and calendars and now it's gone with Asterisk 15.
>>>>> Some bugfixes on recurring events are also included in Asterisk 15.
>>>>>
>>>>> Regards.
>>>>>
>>>>> --
>>>>> Ludovic Gasc (GMLudo)
>>>>>
>>>>> 2018-03-13 20:16 GMT+01:00 Atux At

Re: [asterisk-users] Bank holidays read from file?

[Atux Atux]

Hi. thanks a lot for your reply. i will download the newer libical
software. Could you elaborate on icalendar with google calendar config and
calendar.conf, please?


On Thu, Mar 15, 2018 at 3:00 PM, Ludovic Gasc <gml...@gmail.com> wrote:

> I never use caldav mode, always icalendar with Google Calendar.
>
> BTW, you use old versions of libical, Asterisk and Debian, I recommend you
> to upgrade or install a new server with Debian Stretch: You will have
> Asterisk 13, libical2 and security upgrades.
> Asterisk 11 doesn't have security fixes anymore.
>
> Regards.
>
> --
> Ludovic Gasc (GMLudo)
>
> 2018-03-15 11:28 GMT+01:00 Atux Atux <atuxn...@gmail.com>:
>
>> Hi. Thanks for the idea for calendar, it sounds better. i did not manage
>> to make it work though. i am running debian 8 32 bit with asterisk 11.25.3.
>> I have installed the packages libneon27-dev & libical-dev then in
>> /etc/asterisk the file calendar.conf has the following entries:
>> [Gcalendar]
>>
>> type=caldav
>> url=https://www.google.com/calendar/dav/atuxn...@gmail.com/events/
>> user=atuxn...@gmail.com
>> secret=MySuperDooperPasswd
>> refresh=15
>>
>>
>>
>> then a reload to the system and try to see:
>> PBX> calendar show calendars
>> Calendar Type Status
>>   --
>>
>>
>>
>> there is nothing shown over here. Off course i checked if my calendar is
>> public but i have not an idea why it is not working.
>>
>>
>>
>> On Tue, Mar 13, 2018 at 10:23 PM, Ludovic Gasc <gml...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I recommend you to use calendar module of Asterisk:
>>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Calendaring
>>>
>>> We are using in production since two years with several hundred
>>> calendars, it works pretty well.
>>> However, I strongly recommend you to use the latest stable version of
>>> libical3, because they have fixed a lot of bugs, especially with recurring
>>> events.
>>>
>>> And to use Asterisk 15: We had time to time crashes with Asterisk 13 and
>>> calendars and now it's gone with Asterisk 15.
>>> Some bugfixes on recurring events are also included in Asterisk 15.
>>>
>>> Regards.
>>>
>>> --
>>> Ludovic Gasc (GMLudo)
>>>
>>> 2018-03-13 20:16 GMT+01:00 Atux Atux <atuxn...@gmail.com>:
>>>
>>>> Hi. in my home office i operate my asterisk and have an IVR that has
>>>> the business hours 9-5 and everytime i edit it to load the bank holidays
>>>> (New Years eve, christmas, easter, whatever else). I would like to be able
>>>> to load in the Asterisk's DB or in a file for all the year or years the
>>>> planned holidays. Then it will be read from that file to operate
>>>> accordingly.
>>>> Is there a hint on how to run something like that?
>>>> I am running asterisk 11.
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>> --
>> _
>> -- 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

Re: [asterisk-users] Bank holidays read from file?

[Atux Atux]

Hi. Thanks for the idea for calendar, it sounds better. i did not manage to
make it work though. i am running debian 8 32 bit with asterisk 11.25.3. I
have installed the packages libneon27-dev & libical-dev then in
/etc/asterisk the file calendar.conf has the following entries:
[Gcalendar]

type=caldav
url=https://www.google.com/calendar/dav/atuxn...@gmail.com/events/
user=atuxn...@gmail.com
secret=MySuperDooperPasswd
refresh=15



then a reload to the system and try to see:
PBX> calendar show calendars
Calendar Type Status
  --



there is nothing shown over here. Off course i checked if my calendar is
public but i have not an idea why it is not working.



On Tue, Mar 13, 2018 at 10:23 PM, Ludovic Gasc <gml...@gmail.com> wrote:

> Hi,
>
> I recommend you to use calendar module of Asterisk:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Calendaring
>
> We are using in production since two years with several hundred calendars,
> it works pretty well.
> However, I strongly recommend you to use the latest stable version of
> libical3, because they have fixed a lot of bugs, especially with recurring
> events.
>
> And to use Asterisk 15: We had time to time crashes with Asterisk 13 and
> calendars and now it's gone with Asterisk 15.
> Some bugfixes on recurring events are also included in Asterisk 15.
>
> Regards.
>
> --
> Ludovic Gasc (GMLudo)
>
> 2018-03-13 20:16 GMT+01:00 Atux Atux <atuxn...@gmail.com>:
>
>> Hi. in my home office i operate my asterisk and have an IVR that has the
>> business hours 9-5 and everytime i edit it to load the bank holidays (New
>> Years eve, christmas, easter, whatever else). I would like to be able to
>> load in the Asterisk's DB or in a file for all the year or years the
>> planned holidays. Then it will be read from that file to operate
>> accordingly.
>> Is there a hint on how to run something like that?
>> I am running asterisk 11.
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] Bank holidays read from file?

[Atux Atux]

Hi. in my home office i operate my asterisk and have an IVR that has the
business hours 9-5 and everytime i edit it to load the bank holidays (New
Years eve, christmas, easter, whatever else). I would like to be able to
load in the Asterisk's DB or in a file for all the year or years the
planned holidays. Then it will be read from that file to operate
accordingly.
Is there a hint on how to run something like that?
I am running asterisk 11.
-- 
_
-- 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

[asterisk-users] Asterisk server as TLS/SRTP

[Atux Atux]

Hi. I have an Asterisk Server (A) where it acts as the main gateway to
offer services.
There are different asterisk servers (B -D) that connect as extensions to
the Server A.
I would like to implement TLS and SRTP for these extensions, but have the
non secure as well for other extensions.
for example the extensions 4500-4504 be with TLS/SRTP and the rest be non
secure(ordinary).
Is there a guide on how to implement that please?
I am running asterisk 11.
-- 
_
-- 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

[asterisk-users] Blacklist failed attempts

[Atux Atux]

Hi. I would like to protect my system from failed attempts. I would like to
ask if there is a way to do a blacklist for certain amount of time
consecutive attempts from the same IP. For example if we have an IP that
gets a wrong passwd an it had tried more than 3 times the last 5 minutes,
blacklist it for an hour. I have tried to implement it through fail2ban,
but it doe snot seem to work for my asterisk implementation.
Is there any other way?
-- 
_
-- 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

Re: [asterisk-users] how do i enable call features??

[Atux Atux]

Being honest, i did not manage to make it work. Now whoever calls the
system extensions, does not know if they are on another phone call or away
from the office.

On Tue, Jan 16, 2018 at 12:30 PM, Atux Atux <atuxn...@gmail.com> wrote:

> at the moment i have in each extension in sip.conf the call-limit=2.
> Everytime someone calls that extension and that extension is busy, there is
> not any notification:
> - to the extension that there is a second call
> -to the calling party that this extension is on call. So the calling can
> either wait or hang up.
>
>
> How can i make that happen, please?
>
> On Thu, Jan 11, 2018 at 9:58 AM, Atux Atux <atuxn...@gmail.com> wrote:
>
>> No idea on how to write it in my system.
>>
>> On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkinis...@gmail.com>
>> wrote:
>>
>>> There's some example code in the Dial-Users context of the basic-pbx
>>> samples that might be of use in implementing it.
>>>
>>> They are checking a DEVICE_STATE to see if a phone is BUSY, You could
>>> change it to be a database call or implement custom device states and check
>>> those.
>>>
>>> wrapping your test case in an ExecIF statement that uses the DB_EXISTS
>>> function to see if the database field you are checking is valid so you
>>> don't get errors about non existent entries.
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Funct
>>> ion_DB_EXISTS
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB
>>>
>>> On Wed, Jan 10, 2018 at 11:19 AM, Atux Atux <atuxn...@gmail.com> wrote:
>>>
>>>> That is the general idea. But how do i make it work? is there somewhere
>>>> ready?
>>>>
>>>>
>>>> On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkinis...@gmail.com>
>>>> wrote:
>>>>
>>>>> Define your *72 and *73 extensions in your internal context, Have them
>>>>> set a value in the ASTDB that you then check when dialing your handsets.
>>>>>
>>>>> The same can be done for call forwarding, store a number in the ASTDB
>>>>> and check if it's present, if it is forward the call to that number.
>>>>>
>>>>> On Wed, Jan 10, 2018 at 12:18 AM, Atux Atux <atuxn...@gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi. i am running asterisk 11 and i would like to have features access
>>>>>> codes in my system such as call waiting(all types) (enable/disable), call
>>>>>> forward (enable/disable) and DND. my dialplan is pretty simple and it is
>>>>>> the following
>>>>>>
>>>>>> [DefaultPlan]exten => 
>>>>>> _XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => 
>>>>>> _XX,1,Busy()
>>>>>> exten => _4XX,2,Answer()exten => 
>>>>>> _4XX,3,VoiceMail(${EXTEN}@Office,su)exten => _4XX,4,HangUp()exten => 
>>>>>> _4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten => 
>>>>>> _4XX,104,HangUp()
>>>>>>
>>>>>> i would like to enable/disable call waiting by typing eg. *70/*71
>>>>>> DND for the extension *72 enable, *73 to disable.
>>>>>>
>>>>>> Regarding call waiting, at the moment it is disabled (default value).
>>>>>> Now if an extension is busy, a busy message is send back to the caller. I
>>>>>> would like have the following behavior:
>>>>>> -in the event were the extension is busy, then send a message
>>>>>> indication to the extension and the caller to hear from the SIP provider
>>>>>> the default early media for call waiting due to busy. Then after some
>>>>>> period of time eg 30 secs send busy.
>>>>>> -in the event where the extension is busy, send the early media to
>>>>>> the caller and waiting indication to the extension. If the extension
>>>>>> decides to get the call then get the 2nd call and send the 1st to hear 
>>>>>> moh.
>>>>>>
>>>>>> My phones are mainly softphones (zoiper), a few IP phones and 2
>>>>>> SPA3000 for analog devices.
>>>>>>
>>>>>> could someone help me please with this task, please?
>>>>>>
>>>>>> --
>>>>>> ___

[asterisk-users] asterisk mysql contacts

[Atux Atux]

Hi. i have an asterisk 11 installation that i run in my soho environment.
My system has mysql to store all the cdrs.
I would like make use of the mysql and store numbers and names. eg
+4922123456789 "Atux Null". So when the +4922123456789 calls in my system
the name "Atux Null" will pop up next to the number.
at the moment i have a database called MyNames in mysql that has this
information, but i do not know how to make the dialplan read from this
database.
I would like to ask if there is a way to implement this easily in my
dialplan, please.
-- 
_
-- 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

Re: [asterisk-users] how do i enable call features??

[Atux Atux]

at the moment i have in each extension in sip.conf the call-limit=2.
Everytime someone calls that extension and that extension is busy, there is
not any notification:
- to the extension that there is a second call
-to the calling party that this extension is on call. So the calling can
either wait or hang up.


How can i make that happen, please?

On Thu, Jan 11, 2018 at 9:58 AM, Atux Atux <atuxn...@gmail.com> wrote:

> No idea on how to write it in my system.
>
> On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkinis...@gmail.com>
> wrote:
>
>> There's some example code in the Dial-Users context of the basic-pbx
>> samples that might be of use in implementing it.
>>
>> They are checking a DEVICE_STATE to see if a phone is BUSY, You could
>> change it to be a database call or implement custom device states and check
>> those.
>>
>> wrapping your test case in an ExecIF statement that uses the DB_EXISTS
>> function to see if the database field you are checking is valid so you
>> don't get errors about non existent entries.
>>
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_EXISTS
>>
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB
>>
>> On Wed, Jan 10, 2018 at 11:19 AM, Atux Atux <atuxn...@gmail.com> wrote:
>>
>>> That is the general idea. But how do i make it work? is there somewhere
>>> ready?
>>>
>>>
>>> On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkinis...@gmail.com>
>>> wrote:
>>>
>>>> Define your *72 and *73 extensions in your internal context, Have them
>>>> set a value in the ASTDB that you then check when dialing your handsets.
>>>>
>>>> The same can be done for call forwarding, store a number in the ASTDB
>>>> and check if it's present, if it is forward the call to that number.
>>>>
>>>> On Wed, Jan 10, 2018 at 12:18 AM, Atux Atux <atuxn...@gmail.com> wrote:
>>>>
>>>>> Hi. i am running asterisk 11 and i would like to have features access
>>>>> codes in my system such as call waiting(all types) (enable/disable), call
>>>>> forward (enable/disable) and DND. my dialplan is pretty simple and it is
>>>>> the following
>>>>>
>>>>> [DefaultPlan]exten => 
>>>>> _XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => 
>>>>> _XX,1,Busy()
>>>>> exten => _4XX,2,Answer()exten => 
>>>>> _4XX,3,VoiceMail(${EXTEN}@Office,su)exten => _4XX,4,HangUp()exten => 
>>>>> _4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten => 
>>>>> _4XX,104,HangUp()
>>>>>
>>>>> i would like to enable/disable call waiting by typing eg. *70/*71
>>>>> DND for the extension *72 enable, *73 to disable.
>>>>>
>>>>> Regarding call waiting, at the moment it is disabled (default value).
>>>>> Now if an extension is busy, a busy message is send back to the caller. I
>>>>> would like have the following behavior:
>>>>> -in the event were the extension is busy, then send a message
>>>>> indication to the extension and the caller to hear from the SIP provider
>>>>> the default early media for call waiting due to busy. Then after some
>>>>> period of time eg 30 secs send busy.
>>>>> -in the event where the extension is busy, send the early media to the
>>>>> caller and waiting indication to the extension. If the extension decides 
>>>>> to
>>>>> get the call then get the 2nd call and send the 1st to hear moh.
>>>>>
>>>>> My phones are mainly softphones (zoiper), a few IP phones and 2
>>>>> SPA3000 for analog devices.
>>>>>
>>>>> could someone help me please with this task, please?
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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

[asterisk-users] email when certain numbers are called

[Atux Atux]

Hi. I have an installation of asterisk 11 and i have ssmtp in the system to
send emails. I would like to get informed by email when someone dials a set
of numbers eg international calls or premium numbers with the country. my
dialplan is simple enough and it is the following:

[DefaultPlan]exten =>
_XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten =>
_XX,1,Busy()
exten => _4XX,2,Answer()exten =>
_4XX,3,VoiceMail(${EXTEN}@BranchA,su)exten => _4XX,4,HangUp()exten =>
_4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@BranchA,sb)exten
=> _4XX,104,HangUp()
-- 
_
-- 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

Re: [asterisk-users] how do i enable call features??

[Atux Atux]

No idea on how to write it in my system.

On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkinis...@gmail.com>
wrote:

> There's some example code in the Dial-Users context of the basic-pbx
> samples that might be of use in implementing it.
>
> They are checking a DEVICE_STATE to see if a phone is BUSY, You could
> change it to be a database call or implement custom device states and check
> those.
>
> wrapping your test case in an ExecIF statement that uses the DB_EXISTS
> function to see if the database field you are checking is valid so you
> don't get errors about non existent entries.
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_EXISTS
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB
>
> On Wed, Jan 10, 2018 at 11:19 AM, Atux Atux <atuxn...@gmail.com> wrote:
>
>> That is the general idea. But how do i make it work? is there somewhere
>> ready?
>>
>>
>> On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkinis...@gmail.com>
>> wrote:
>>
>>> Define your *72 and *73 extensions in your internal context, Have them
>>> set a value in the ASTDB that you then check when dialing your handsets.
>>>
>>> The same can be done for call forwarding, store a number in the ASTDB
>>> and check if it's present, if it is forward the call to that number.
>>>
>>> On Wed, Jan 10, 2018 at 12:18 AM, Atux Atux <atuxn...@gmail.com> wrote:
>>>
>>>> Hi. i am running asterisk 11 and i would like to have features access
>>>> codes in my system such as call waiting(all types) (enable/disable), call
>>>> forward (enable/disable) and DND. my dialplan is pretty simple and it is
>>>> the following
>>>>
>>>> [DefaultPlan]exten => 
>>>> _XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => 
>>>> _XX,1,Busy()
>>>> exten => _4XX,2,Answer()exten => _4XX,3,VoiceMail(${EXTEN}@Office,su)exten 
>>>> => _4XX,4,HangUp()exten => _4XX,102,Answer()exten => 
>>>> _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten => _4XX,104,HangUp()
>>>>
>>>> i would like to enable/disable call waiting by typing eg. *70/*71
>>>> DND for the extension *72 enable, *73 to disable.
>>>>
>>>> Regarding call waiting, at the moment it is disabled (default value).
>>>> Now if an extension is busy, a busy message is send back to the caller. I
>>>> would like have the following behavior:
>>>> -in the event were the extension is busy, then send a message
>>>> indication to the extension and the caller to hear from the SIP provider
>>>> the default early media for call waiting due to busy. Then after some
>>>> period of time eg 30 secs send busy.
>>>> -in the event where the extension is busy, send the early media to the
>>>> caller and waiting indication to the extension. If the extension decides to
>>>> get the call then get the 2nd call and send the 1st to hear moh.
>>>>
>>>> My phones are mainly softphones (zoiper), a few IP phones and 2 SPA3000
>>>> for analog devices.
>>>>
>>>> could someone help me please with this task, please?
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>>
>>>
>>> --
>>> A human being should be able to change a diaper, plan an invasion,
>>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>>> manure, program a computer, cook a tasty meal, fight efficiently, die
>>> gallantly. Specialization is for insects.
>>> ---Heinlein
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] how do i enable call features??

[Atux Atux]

That is the general idea. But how do i make it work? is there somewhere
ready?


On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkinis...@gmail.com>
wrote:

> Define your *72 and *73 extensions in your internal context, Have them set
> a value in the ASTDB that you then check when dialing your handsets.
>
> The same can be done for call forwarding, store a number in the ASTDB and
> check if it's present, if it is forward the call to that number.
>
> On Wed, Jan 10, 2018 at 12:18 AM, Atux Atux <atuxn...@gmail.com> wrote:
>
>> Hi. i am running asterisk 11 and i would like to have features access
>> codes in my system such as call waiting(all types) (enable/disable), call
>> forward (enable/disable) and DND. my dialplan is pretty simple and it is
>> the following
>>
>> [DefaultPlan]exten => _XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten 
>> => _XX,1,Busy()
>> exten => _4XX,2,Answer()exten => _4XX,3,VoiceMail(${EXTEN}@Office,su)exten 
>> => _4XX,4,HangUp()exten => _4XX,102,Answer()exten => 
>> _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten => _4XX,104,HangUp()
>>
>> i would like to enable/disable call waiting by typing eg. *70/*71
>> DND for the extension *72 enable, *73 to disable.
>>
>> Regarding call waiting, at the moment it is disabled (default value). Now
>> if an extension is busy, a busy message is send back to the caller. I would
>> like have the following behavior:
>> -in the event were the extension is busy, then send a message indication
>> to the extension and the caller to hear from the SIP provider the default
>> early media for call waiting due to busy. Then after some period of time eg
>> 30 secs send busy.
>> -in the event where the extension is busy, send the early media to the
>> caller and waiting indication to the extension. If the extension decides to
>> get the call then get the 2nd call and send the 1st to hear moh.
>>
>> My phones are mainly softphones (zoiper), a few IP phones and 2 SPA3000
>> for analog devices.
>>
>> could someone help me please with this task, please?
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] how do i enable call features??

[Atux Atux]

Hi. i am running asterisk 11 and i would like to have features access codes
in my system such as call waiting(all types) (enable/disable), call forward
(enable/disable) and DND. my dialplan is pretty simple and it is the
following

[DefaultPlan]exten =>
_XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten =>
_XX,1,Busy()
exten => _4XX,2,Answer()exten =>
_4XX,3,VoiceMail(${EXTEN}@Office,su)exten => _4XX,4,HangUp()exten =>
_4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten
=> _4XX,104,HangUp()

i would like to enable/disable call waiting by typing eg. *70/*71
DND for the extension *72 enable, *73 to disable.

Regarding call waiting, at the moment it is disabled (default value). Now
if an extension is busy, a busy message is send back to the caller. I would
like have the following behavior:
-in the event were the extension is busy, then send a message indication to
the extension and the caller to hear from the SIP provider the default
early media for call waiting due to busy. Then after some period of time eg
30 secs send busy.
-in the event where the extension is busy, send the early media to the
caller and waiting indication to the extension. If the extension decides to
get the call then get the 2nd call and send the 1st to hear moh.

My phones are mainly softphones (zoiper), a few IP phones and 2 SPA3000 for
analog devices.

could someone help me please with this task, please?
-- 
_
-- 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

Re: [asterisk-users] asterisk name in mysql

[Atux Atux]

I will try to reinstall everything according to your instructions and i
will come back. I might need a couple of weeks due to a business trip though

Στις 24 Απρ 2017 6:54 μ.μ., ο χρήστης "John Kiniston" <
johnkinis...@gmail.com> έγραψε:

> Well, My suggestion was to use FUNC_ODBC but instead you went with
> APP_MYSQL which has been depricated.
>
> Did you compile APP_MYSQL? It's not enabled by default.
>
> On Sat, Apr 22, 2017 at 1:25 PM, Atux Atux <atuxn...@gmail.com> wrote:
>
>> Thanks a lot for the reply.
>> I did follow that already, but i do have a problem. Here is my
>> extensions.conf part for that particular number
>> exten => 6912345678,1,Answer()
>> exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd
>> asterisk)
>>
>> and here is the error i am getting
>> [Apr 22 23:20:29] WARNING[9725][C-0002]: pbx.c:4991
>> pbx_extension_helper: No application 'MYSQL' for extension (IncomingDial,
>> 6951921078, 2)
>>   == Spawn extension (DialIn, 6912345678, 2) exited non-zero on
>> 'Dongle/dongle0-010002'
>>
>>
>> Any ideas please?
>>
>
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] asterisk name in mysql

[Atux Atux]

Thanks a lot for the reply.
I did follow that already, but i do have a problem. Here is my
extensions.conf part for that particular number
exten => 6912345678,1,Answer()
exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd asterisk)
exten => 6912345678,n,MYSQL(Query resultid ${connid} SET NAMES utf8)
exten => 6912345678,n,GotoIf($["${connid}" = ""]?nodb)
exten => 6912345678,n,MYSQL(Query resultid ${connid} SELECT displayname
FROM root WHERE phonenumber="${CALLERID(num)}" LIMIT 1)
exten => 6912345678,n,MYSQL(Fetch fetchid ${resultid} displayname)
exten => 6912345678,n,MYSQL(Clear ${resultid})
exten => 6912345678,n,Set(CALLERID(name)=${displayname})
exten => 6912345678,n,MYSQL(Disconnect ${connid})
exten => 6912345678,n(nodb),NoOp(DoneDB)
exten => 6912345678,n,Dial(SIP/450/451,20)
exten => 6912345678,n,VoiceMail(450@Office,su)
exten => 6912345678,n,Busy(3)

6912345678 is my DID



and here is the error i am getting
[Apr 22 23:20:29] WARNING[9725][C-0002]: pbx.c:4991
pbx_extension_helper: No application 'MYSQL' for extension (IncomingDial,
6951921078, 2)
  == Spawn extension (DialIn, 6912345678, 2) exited non-zero on
'Dongle/dongle0-010002'


Any ideas please?


On Fri, Apr 21, 2017 at 10:22 PM, John Kiniston <johnkinis...@gmail.com>
wrote:

> You can use func_odbc to do this.
>
> https://wiki.asterisk.org/wiki/display/AST/Getting+
> Asterisk+Connected+to+MySQL+via+ODBC2
>
> There is a good chapter in the Asterisk book about using ODBC for
> hotdesking that may help you understand ODBC as well.
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-
> html-chunk/getting_funky.html
>
> On Fri, Apr 21, 2017 at 12:12 PM, Atux Atux <atuxn...@gmail.com> wrote:
>
>> hi. currently i am running the phonebook in astdb with
>>
>>
>> *database put cidname 0123456789 "name_surname"*
>> and i retrive it with
>>
>>
>> *exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
>> Now, my system has mysql and i got all my contacts in there in a database
>> is called *asterisk *and a table called *addressbook**. *password of the
>> mysql is
>>
>> *whateverpasswd*
>> how do i alter the above section of the extensions.conf code to query
>> mysql everytime there is a call, please?
>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] asterisk name in mysql

[Atux Atux]

hi. currently i am running the phonebook in astdb with


*database put cidname 0123456789 "name_surname"*
and i retrive it with


*exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
Now, my system has mysql and i got all my contacts in there in a database
is called *asterisk *and a table called *addressbook**. *password of the
mysql is

*whateverpasswd*
how do i alter the above section of the extensions.conf code to query mysql
everytime there is a call, please?
-- 
_
-- 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

Re: [asterisk-users] asterisk as non root

[Atux Atux]

the output of ls -l is
root@pbx: ~ $ ls -l /var/run/asterisk/asterisk.ctl
srwxr-xr-x 1 asterisk asterisk 0 Apr 20 19:47 /var/run/asterisk/asterisk.ctl
root@pbx: ~ $


On Thu, Apr 20, 2017 at 7:46 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:
>
> > root@PBX: /var/www/html $ /etc/init.d/asterisk start
> > [ ok ] Starting asterisk (via systemctl): asterisk.service.
>
> I'm somewhat puzzled that your root-user prompt is "$"
> instead of the more normal "#", but never mind...
>
> > root@PBX: /var/www/html $ ps aux | grep asterisk
> > asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49
> /usr/sbin/asterisk -U asterisk -G asterisk
>
> So, the first column of that output shows you that asterisk is
> running as the user "asterisk".
>
> On my Debian system I only have "-U asterisk" without the "-G asterisk".
>
> > root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep
> asterisk
>
> ...and the grep command was run by "root"
>
> > root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
> > Privilege escalation protection disabled!
> > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> > Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use
> 'asterisk
> > -r' to connect.
>
> Who does "ls -l" show you that file /var/run/asterisk/asterisk.ctl
> is owned by?
>
> On my machine it's:
>
> srwxrwx--- 1 asterisk asterisk 0 Apr 11 10:32
> /var/run/asterisk/asterisk.ctl
>
>
> Antony.
>
> --
> There's a good theatrical performance about puns on in the West End.  It's
> a play on words.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] asterisk as non root

[Atux Atux]

root@PBX: /var/www/html $ /etc/init.d/asterisk start
[ ok ] Starting asterisk (via systemctl): asterisk.service.
root@PBX: /var/www/html $ ps aux | grep asterisk
asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49
/usr/sbin/asterisk -U asterisk -G asterisk
root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep
asterisk
root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
-r' to connect.
root@PBX: /var/www/html $



On Thu, Apr 20, 2017 at 1:36 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:
>
> > Hi. thanks a lot for your replies. I did stop the services and i did
> issued
> > the  the "chown" and "chmod" commands listed in the guide.
> > It is necessary to compile it, instead if using the apt-get version
> > What am i missing?
>
> Let's go back to basics for a moment - you say this is a Debian system -
> in my
> experience Debian already runs Asterisk as the "asterisk" user and not as
> root, so let's see what you have.
>
> 1. Start Asterisk (probably using "/etc/init.d/asterisk start", or maybe
> "service asterisk start")
>
> 2. Check who it's running as: "ps aux | grep asterisk"
>
>
> Antony.
>
>
> --
> What makes you think I know what I'm talking about?
> I just have more O'Reilly books than most people.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] asterisk as non root

[Atux Atux]

Hi. thanks a lot for your replies. I did stop the services and i did issued
the  the "chown" and "chmod" commands listed in the guide.
It is necessary to compile it, instead if using the apt-get version
What am i missing?



On Wed, Apr 19, 2017 at 10:47 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:
>
> > Hi.
> > Here is the output of the command
> >
> > root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
> >
> > drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
> >
> > drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
> >
> > -rwxr-xr-x 1 root root 9719880 Apr 19 17:27 /usr/src/asterisk-11.25.1/
> main/asterisk
> >
> > drwxrwxr-x 3 1013 users 4096 Apr 19 16:56 /usr/src/asterisk-11.25.1/
> include/asterisk
> >
> > -rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk
>
> Okay, those look reasonable to me - however I'm surprised at some which
> are missing:
>
> /var/log/asterisk
> /var/spool/asterisk
> /var/run/asterisk
>
> Did you *stop* Asterisk before trying to change it to run as non-root?
>
> I think Tzafrir Cohen's comments are very well worth following.
>
>
> Antony.
>
> --
> "There has always been an underlying argument that we should open up our
> source code more broadly. The fact is that we are learning from open source
> and we are opening our code more broadly through Shared Source.
>
> Is there value to providing source code? The answer is unequivocally yes."
>
>  - Jason Matusow, head of Microsoft's Shared Source Program, in response
> to leaks of Windows source code on the Internet.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] asterisk as non root

[Atux Atux]

Hi.
Here is the output of the command

root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
-rwxr-xr-x 1 root root 9719880 Apr 19 17:27
/usr/src/asterisk-11.25.1/main/asterisk
drwxrwxr-x 3 1013 users 4096 Apr 19 16:56
/usr/src/asterisk-11.25.1/include/asterisk
-rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk
root@pbx: ~ $


On Wed, Apr 19, 2017 at 5:03 PM, Tzafrir Cohen <tzafrir.co...@xorcom.com>
wrote:

> On Wed, Apr 19, 2017 at 04:44:39PM +0300, Atux Atux wrote:
> > hello there. i am running debian 8 in my swerver and i would like to run
> > asterisk as non root.
>
> The Asterisk package included with Debian already does that. Why not
> have a look at it?
>
> > i did follow the
> > https://www.voip-info.org/wiki-Asterisk+non-root without any success.
> when
> > i issue
> > root@PBX: ~ $ asterisk -U asterisk -G asterisk
>
> The options -U and -G are for the case of running Asterisk as root and
> having Asterisk change user and group afterwards. There are a number of
> options that only work that way (real-time priority, special socket
> permissions, IIRC).
>
> Alternatively you can use other mans to change to that user (--chuid or
> start-stop-daemon or User: and Group: in a systemd service file, or
> whatever). And then you don't need those options.
>
> > Privilege escalation protection disabled!
> > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
>
> Read that text. But it is irrelevant for your situation.
>
> > Unable to access the running directory (Permission denied). Changing to
> '/'
> > for compatibility.
>
> /root is not accessible by the user asterisk. This is mostly harmless,
> but not if you want to have core files (see also -g) and maybe a few
> other minor things.
>
> > Asterisk already running on /var/run/asterisk/asterisk.ctl. Use
> 'asterisk
> > -r' to connect.
>
> Because you already ran that command before. Or already have the system
> copy of asterisk running. Or whatever.
>
> Reading error messages helps.
>
> --
>Tzafrir Cohen
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] asterisk as non root

[Atux Atux]

hello there. i am running debian 8 in my swerver and i would like to run
asterisk as non root. i did follow the
https://www.voip-info.org/wiki-Asterisk+non-root without any success. when
i issue
root@PBX: ~ $ asterisk -U asterisk -G asterisk
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Unable to access the running directory (Permission denied). Changing to '/'
for compatibility.
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk
-r' to connect.
root@PBX: ~ $


any ideas on how to fix that please?
-- 
_
-- 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

Re: [asterisk-users] restart system from extension

[Atux Atux]

Could you give some more details please?

Στις 6 Απρ 2017 8:25 μ.μ., ο χρήστης "Tzafrir Cohen" <
tzafrir.co...@xorcom.com> έγραψε:

> On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote:
> > hi. i would like to be able to reboot the system from my extension. is
> that
> > possible? if yes, how?
>
> System('sudo /sbin/reboot')
>
> You need to allow that in a sudoers file, of course. This may or may not
> be a good idea.
>
> There are a host of other methods to permit unplivilidged users /
> processes to run do specific priviliged actions.
>
> --
>Tzafrir Cohen
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] restart system from extension

[Atux Atux]

hi. i would like to be able to reboot the system from my extension. is that
possible? if yes, how?
-- 
_
-- 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

Re: [asterisk-users] feature codes

[Atux Atux]

hi. thanks for the reply. when you say configure them in features.conf?

On Thu, Apr 6, 2017 at 11:57 AM, Marcelo Terres <mhter...@gmail.com> wrote:

> You can configure the features in the features.conf file, but some
> features like DND and call forward are not available, so, or you use
> the SIP client own functionalities for that (if available), or you
> will have to develop your own features.
>
> Regards,
> Marcelo H. Terres <mhter...@gmail.com>
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 6 April 2017 at 08:46, Atux Atux <atuxn...@gmail.com> wrote:
> > hi.
> >
> > i am running asterisk 11 and i am stuck with the feature codes. how do i
> > setup them.
> > Now the system has.
> >
> > PBX*CLI> features show
> > Builtin Feature Default Current
> > --- --- ---
> > Pickup *8 *8
> > Blind Transfer # #
> > Attended Transfer
> > One Touch Monitor
> > Disconnect Call * *
> > Park Call
> > One Touch MixMonitor
> >
> > Dynamic Feature Default Current
> > --- --- ---
> > (none)
> >
> > Feature Groups:
> > ---
> > (none)
> >
> > Call parking (Parking lot: default)
> > 
> > Parking extension : 700
> > Parking context : parkedcalls
> > Parked call extensions: 701-750
> > Parkingtime : 45000 ms
> > Comeback to origin : yes
> > Comeback context : parkedcallstimeout (comebacktoorigin=yes, not used)
> > Comeback dial time : 30
> > MusicOnHold class : default
> > Enabled : Yes
> > PBX*CLI>
> >
> > My extensions.conf is:
> > exten => _2X,1,Dial(SIP/CYTA/${EXTEN})
> > exten => _2X,1,Busy()
> > exten => _69,1,Dial(SIP/voda/${EXTEN})
> > exten => _69,1,Busy()
> > [code]
> >
> > I would like to be able to transfer calls, blind/attended transfer, call
> > forward, DND. I would appreciate any help available please.
> >
> >
> > --
> > _
> > -- 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
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] feature codes

[Atux Atux]

hi.

i am running asterisk 11 and i am stuck with the feature codes. how do i
setup them.
Now the system has.

PBX*CLI> features show
Builtin Feature Default Current
--- --- ---
Pickup *8 *8
Blind Transfer # #
Attended Transfer
One Touch Monitor
Disconnect Call * *
Park Call
One Touch MixMonitor

Dynamic Feature Default Current
--- --- ---
(none)

Feature Groups:
---
(none)

Call parking (Parking lot: default)

Parking extension : 700
Parking context : parkedcalls
Parked call extensions: 701-750
Parkingtime : 45000 ms
Comeback to origin : yes
Comeback context : parkedcallstimeout (comebacktoorigin=yes, not used)
Comeback dial time : 30
MusicOnHold class : default
Enabled : Yes
PBX*CLI>

*My extensions.conf is:*
exten => _2X,1,Dial(SIP/CYTA/${EXTEN})
exten => _2X,1,Busy()
exten => _69,1,Dial(SIP/voda/${EXTEN})
exten => _69,1,Busy()
[code]

I would like to be able to transfer calls, blind/attended transfer, call
forward, DND. I would appreciate any help available please.
-- 
_
-- 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

[asterisk-users] sms from softphones

[Atux Atux]

hi. in my asterisk i do have a usb 3g dongle, that i am using it for GSM
calls and sms.
At the moment all incoming sms is going to email. outgoing sms is through
the asterisk console: dongle sms dongle0 mobile_number Hello
I would like to ask if it is possible to use my softphones (zoiper) to send
sms through the dongle. If yes, how?
-- 
_
-- 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

[asterisk-users] gcontacts to asterisk

[Atux Atux]

hello everyone. i am looking to automate the management of contacts to my
system (debian 8, with asterisk 11). at the moment i do create the astdb
with database put cidname.
I have searched a bit i have found the google contacts integration
https://zmonkey.org/blog/content/google-contacts-asterisk-caller-id
the problem is that it does not run at all.
my system also has mysql running that i could use.
has anyone any working solution for google contacts in the system, please?
-- 
_
-- 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