Hello,
I have an Asterisk server with a public IP address and a bunch of clients.
Most of my clients are behind NATs (sometimes two clients are behind the
same NAT i.e in the same private network). I want to use ICE so that the
clients behind the same NAT can send RTP traffic directly to each
On Monday 21 Apr 2014, Nick Cameo wrote:
Hello Everyone,
We are looking for a simple open source auto dialer with polling
capabilities. What we would like is a program that we can upload
leads to, and have asterisk:
i) Dial numbers
ii) Play pre-recorded
iii) If user presses one, forward
pls check http://www.newfies-dialer.org/. it may be help you.
Regards
M.Emran
Web: www.e-softbilling.com
www.isoftswitch.com
www.sipmobiledialer.com
www.shop24.com.bd
www.ecitizen.com.bd
--
On 22/4/14 11:44 am, A J Stiles wrote:
Firstly, be warned: Are you sure that is even legal to do in your
jurisdiction? You could be setting yourself up for a hefty fine! Check
applicable local laws before proceeding.
This. I'm glad someone else thought it worth mentioning as well :-)
Even
Our Wombat is not open source but is free to use for small systems and
will be very trivial to set up for such a task. And if you ever need
to grow up, you're covered. :)
l.
2014-04-21 19:45 GMT+02:00 Nick Cameo sym...@gmail.com:
Hello Everyone,
We are looking for a simple open source auto
2014-04-22 3:39 GMT-03:00 binary dreamer dreamer.bin...@gmail.com:
hello there.
sorry if I bring conversation to a different level, but I would suggest you
a cheaper solution.
just get a 3G dongle from the list of http://wiki.e1550.mobi
it is a VERY reliable solution and I have been using it
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
--
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 tone (indications.conf), which works quite well. However the
generated signal is quite loud at the client side (in comparison to the following speech ).
Has anyone here used Asterisk inside a WatchGuard firewall, talking via
the WatchGuard SIP Application Layer Gateway to an outside SIP service?
I have a customer doing just that, and I am 100% convinced there is a bug
in the ALG regarding the media port number it inserts into the SDP when
it
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
On 22 April 2014 16:24, Tony Mountifield t...@softins.co.uk wrote:
Has anyone here used Asterisk inside a WatchGuard firewall, talking via
the WatchGuard SIP Application Layer Gateway to an outside SIP service?
I have a customer doing just that, and I am 100% convinced there is a bug
in the
In article
CAHE6+j3hb5d8mJfY69F73TVwZus9ZAQrDakt4+iW+tx58_uZ=g...@mail.gmail.com,
Ishfaq Malik i...@pack-net.co.uk wrote:
On 22 April 2014 16:24, Tony Mountifield t...@softins.co.uk wrote:
Has anyone here used Asterisk inside a WatchGuard firewall, talking via
the WatchGuard SIP
I would be very surprised is anyone uses WatchGuard SIP ALG. For the past 12
years the advice has always been Disable SIP ALG and let Asterisk do the NAT
fixup itself on any firewall, regardless of brand.I wish you the best of
luck.
-Original Message-
From:
In article
616B4ECE1290D441AD56124FEBB03D0818EB7AE075@mailserver2007.nyigc.globe,
Eric Wieling ewiel...@nyigc.com wrote:
I would be very surprised is anyone uses WatchGuard SIP ALG. For the
past 12 years the advice has always been Disable SIP ALG and let
Asterisk do the NAT fixup itself on
Hello,
I know this is an out of date technology, but I have a need to use
it to reprogram some old phones.
I have managed to get Asterisk and a Digium board to upload scripts
to the phones using the asterisk.adsi example, but trying to make my own
scripts based on the example file
Hello Everyone,
Thank you all for your response. The people I am doing it for run a
non-profit charity,
and are legally able to reach out to their customers. I will wire it
up to the DNC
however, for starters, I would like to get asterisk to:
i) Iterate through a list of numbers
ii) Play a
I can vouch for newfies, but its not asterisk and there is some learning
curve, but comes with lots of features.
-Jai
www.didforsale.com
SIP Trunking Simplified
On Tue, Apr 22, 2014 at 2:54 PM, Nick Cameo sym...@gmail.com wrote:
Hello Everyone,
Thank you all for your response. The people I
On 4/22/2014 5:54 PM, Nick Cameo wrote:
Hello Everyone,
Thank you all for your response. The people I am doing it for run a
non-profit charity,
and are legally able to reach out to their customers. I will wire it
up to the DNC
however, for starters, I would like to get asterisk to:
i) Iterate
That's about as simple as it gets.
A call file that goes to the dialplan.
A dialplan that consists of Read (which would play the message) followed a
GotoIf into a mailbox (either voicemail or Dial() to an external number).
One hint for doing unattended dialing like this, make sure you're
19 matches
Mail list logo
