0/2018 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 strin
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
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
ed #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
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 cal
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
h 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
> &g
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, p
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 w
>
> This should work:
> [call-file-test]
> Exten => 10,1,Answer
> same => ConfBridge(100)
>
> On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux wrote:
>
>> Hi. in my system i have a conference room where someone can call it eg
>> 698 dial the PIN eg 123
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
acces
; 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 :
>
>> Hi. thanks a lot for your reply. i will dow
udovic Gasc (GMLudo)
>
> 2018-03-15 11:28 GMT+01:00 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 &
: 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 :
&g
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 ho
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 examp
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
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 wrote:
> at the moment i have in each extension in sip.conf the call-limit=2.
> Eve
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
or hang up.
How can i make that happen, please?
On Thu, Jan 11, 2018 at 9:58 AM, Atux Atux wrote:
> No idea on how to write it in my system.
>
> On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston
> wrote:
>
>> There's some example code in the Dial-Users context of the
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 =>
rs 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 wrote:
>
>> That is the general idea. But how do i make
gt; 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 wrote:
>
>> Hi. i am running asterisk 11 and i would like to have features access
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/${EXTEN},120,
DBC but instead you went with
> APP_MYSQL which has been depricated.
>
> Did you compile APP_MYSQL? It's not enabled by default.
>
> On Sat, Apr 22, 2017 at 1:25 PM, Atux Atux wrote:
>
>> Thanks a lot for the reply.
>> I did follow that already, but i do have
DBC2
>
> There is a good chapter in the Asterisk book about using ODBC for
> hotdesking that may help you understand ODBC as well.
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-
> html-chunk/getting_funky.html
>
> On Fri, Apr 21, 2017 at 12:12 PM, Atux Atux wrot
hi. currently i am running the phonebook in astdb with
*database put cidname 0123456789 "name_surname"*
and i retrive it with
*exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
Now, my system has mysql and i got all my contacts in there in a database
is called *asterisk
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 n
x27;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
ny 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 r
sk-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
wrote:
> On Wed, Apr 19, 2017 at 04:44:39PM +0300, At
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://wik
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
&g
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://
es
>
>
> On 6 April 2017 at 08:46, Atux Atux 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 Fe
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
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
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-
38 matches
Mail list logo