On Wed, 2017-05-10 at 12:56 +0200, Frank Vanoni wrote:
> exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC)
>
> exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB)
Whoops... sorry for the typo (in the hurry of copy & paste)!
exten => 2001,1,Dial(SIP/deviceA/deviceB/deviceC)
exten =>
Dear Digium List
First of all, I thank all of you for all the replies and the interesting
suggestions. I thank you very much. I can only learn from people like
you. :-)
I will remember all the different solutions for a future use in other
scenarios.
On Mon, 2017-05-08 at 16:35 +0200, Frank
Hello,
I am new to Asterisk, so please bear with me.
I have made a success installation from source of Asterisk 14.4.0 on
Debian Jessie (8.7). And I am running the Asterisk server, with several
extensions and dialplans, all working well.
However I am struggling to get app_jack to run.
In
On Wednesday 10 May 2017, andre castro wrote:
> Hello,
> I am new to Asterisk, so please bear with me.
> I have made a success installation from source of Asterisk 14.4.0 on
> Debian Jessie (8.7). And I am running the Asterisk server, with several
> extensions and dialplans, all working well.
>
>
Rather than that, if you're looking for a phone solution - as part of the
customer contract, install an IP phone that registers with your system (use
a VPN tunnel to your phone system). Think of it like a "red-phone"
hotline. You own the phone, and you physically install it and it only
talks to
It's probably not practical to have them answering the client's telephone!
At a lot of sites, incoming calls would be handled by auto attendant,
diverted to answering service, etc.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Since the callback happens immediately after hangning up, the risk of
answering a call that isn't theirs is minimal.
For those sites that divert their incoming calls to a PBX or answering
machine, you could have some config/database that excepts these sites from
callback verification.
(which means
On 2017-05-10 04:15 PM, Sebastian Nielsen wrote:
The thing is then to be able to record which IP is the client, but if your
services are ordered by the client via some web form, you could have that IP
be recorded as "client IP" and the employee must check in/check out from
that IP.
IPs change.
On Wed, Apr 26, 2017 at 06:25:43PM +0200, Daniel Tryba wrote:
> Whoever when a terminating call comes in from the uplink provider, a
> sip request is send to a redirector. The redirector has
> redirect_method=uri_core configured (the only method that works for
> me).
[...]
> The request now gets
Thanks J.
It didn't work.
On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote:
> On Wednesday 10 May 2017, andre castro wrote:
>> Hello,
>> I am new to Asterisk, so please bear with me.
>> I have made a success installation from source of Asterisk 14.4.0 on
>> Debian Jessie (8.7). And I am
Thanks Joshua,
rerun ./configure did the job.
On 05/10/2017 02:48 PM, Joshua Colp wrote:
> On Wed, May 10, 2017, at 09:33 AM, andre castro wrote:
>> Thanks J.
>> It didn't work.
>>
>> On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote:
>>> On Wednesday 10 May 2017, andre castro wrote:
On Wednesday 10 May 2017, andre castro wrote:
> Indeed. apt-get install libjack-dev libresample-dev were not installed.
> libjack-dev libresample-dev , so I installed.
> In the installation of libresample-dev apt-get selected
> 'libresample1-dev' instead of 'libresample-dev'. Not sure if that is a
On Wed, May 10, 2017, at 09:33 AM, andre castro wrote:
> Thanks J.
> It didn't work.
>
> On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote:
> > On Wednesday 10 May 2017, andre castro wrote:
> >> Hello,
> >> I am new to Asterisk, so please bear with me.
> >> I have made a success installation
>>> I ask my SIP provider to include more headers to show the real ANI? What
>>> would that service be
>>> called?
If it's anything like a PRI provider, I've been told they only way to get true
CID, in those instances, would be to provide a 1-800 number (US) for them to
call. Then you'd get
On Wed, May 10, 2017 at 10:11 AM, Steve Edwards
wrote:
> I have a 'time and attendance' application. Think janitorial or security
> kind of thing where an employee goes from location to location.
>
> They're supposed to 'clock in' when they get to a site using a phone
On Wednesday 10 May 2017, Steve Edwards wrote:
> I have a 'time and attendance' application. Think janitorial or security
> kind of thing where an employee goes from location to location.
>
> They're supposed to 'clock in' when they get to a site using a phone at
> that site to prove they're
You have an unusual situation--you suspect caller ID spoofing by a known
person.
Under the Truth in Caller ID Act, FCC rules prohibit any person or entity
from transmitting misleading or inaccurate caller ID information with the
intent to defraud, cause harm, or wrongly obtain anything of value.
It's approximately impossible with current infrastructure.
https://transition.fcc.gov/cgb/Robocall-Strike-Force-Final-Report.pdf
Adam Goldberg
AGP, LLC
+1-202-507-9900
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
I have a 'time and attendance' application. Think janitorial or security
kind of thing where an employee goes from location to location.
They're supposed to 'clock in' when they get to a site using a phone at
that site to prove they're there.
Some employees have discovered 'fake caller ID'
On Wed, 10 May 2017, J Montoya or A J Stiles wrote:
Presumably your staff carry mobile phones. What about an app that gets
the ID of the cell tower to which it is connected, and passes it and the
SIM number in a HTTP request to a server you control?
The problem is that they are supposed to
Use a callback.
So when clocking in/out, they will hear a random 4 digit PIN, like "Enter
four, three, six, eight at the callback".
After they hangup, the phone will ring, and then they will have confirm with
the 4 digit PIN.
If they arent in presence: the phone at the site will ring, and the
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