similar to: withheld caller id

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

so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote: > after adding the ww: > root at Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5 -- Executing > [9211123456 at AllCalls:1] Goto("SIP/500-00000003",

>>> > 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

No idea on how to write it in my system. On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkiniston at 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

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 => _XXXXXXXXXX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => _XXXXXXXXXX,1,Busy() exten => _4XX,2,Answer()exten =>

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 <johnkiniston at 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

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

Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) -

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

hi. i would like to be able to reboot the system from my extension. is that possible? if yes, how? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170406/55c7bee2/attachment.html>

Heres a weird one... Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks up the number in the astdb and puts an name to it. No real magic there, and it works well. Same macro also has parameter passed in to put a prefix on the name - this is set in the DDI handling and is dependent on the number called and allows phone users to see which number was called (company

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

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}" =

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 <gmludo at 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

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/atuxnull at gmail.com/events/ user=atuxnull 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 at 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

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

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

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.Stone at asterisk.open.source.it> wrote: > On Wednesday 19 April 2017 at 18:48:29, Atux

root at PBX: /var/www/html $ /etc/init.d/asterisk start [ ok ] Starting asterisk (via systemctl): asterisk.service. root at 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/0 S+ 17:30 0:00 grep asterisk root at PBX: /var/www/html $