Hi,
I use Asterisk to create the dial tone (indications.conf), which works
quite well. However the generated signal is quite loud at the client side
(in comparison to the following speech ).
Is there an option to modify the volume?
---
Dennis Guse
I let Asterisk generate the ringtone: DIAL(SIP/XX, 'r')...
---
Dennis Guse
On Tue, Apr 22, 2014 at 4:47 PM, jg webaccounts...@jgoettgens.de wrote:
The call invitation is only signaled in most cases. You need to check
the settings of your phones.
Hi,
I use Asterisk to create the dial
})
How to implement the same functionality using pbx_lua?
Details: Asterisk 11.7 on Ubuntu 14.04
Kind regards
Dennis Guse
Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.g
Got it:
extensions = {
[macro-test] = {
[s] = function(c, e)
app.verbose(This is my macro)
end;
};
default = {
[_X] = function(c, e)
app.dial(SIP/00, nil, mM(test))
end;
};
};
---
Dennis Guse
On Fri, Jun 6, 2014 at 6:49 PM, George Joseph george.jos...@fairview5.com
wrote:
On Fri, Jun 6
You could try either the predial-handler or the dial-macro M.
---
Dennis Guse
On Wed, Jul 2, 2014 at 3:06 PM, Joshua Colp jc...@digium.com wrote:
Anurag Rana wrote:
Hi All,
Kia ora,
I am trying to execute some AGI script no matter what extension is called.
There is 'h' extension
Sound like chan_sip was not build.
Just a guess: check that openssl-dev is available
---
Dennis Guse
On Thu, Jul 3, 2014 at 12:01 PM, Andrew Colin and...@vsave.co.za wrote:
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c
On VoIP echo cancellation is basically: hope that the client is doing AND
is doing it well.
In the best case each client uses a knowledge about his hardware
(microphone, speaker, distance etc.).
---
Dennis Guse
On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez emilianovazq...@gmail.com
wrote
configuration I put it like this:
[general]
register = SIP1
[SIP1]
...
register = SIP2
[SIP2]
...
And then the second register is ignored as it is not in [general].
However, no error messages are thrown...
Best regards and a happy weekend!
---
Dennis Guse
---
Dennis Guse
On Wed, Mar 18, 2015 at 10:19 PM
obeying the first register statement.
sip show registry only reports the first entry and if I reorder them,
this effect stays the same.
Did something changed recently in the parsing code for sip.conf or so?
---
Dennis Guse
