Hello everybody,
I have a problem with the application ConfBridge.
I have to create an IVR relief, this is my idea:
- The person in danger call a number (969696);
- Are created call file to call the doctors;
- The person enters a danger confbridge;
If the doctors answer the phone, enter the
I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good
stability.
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Mauricio Tavares
Envoyé : mercredi 16 octobre 2013 15:49
À : Asterisk
On 10/17/2013 09:47 AM, Alban Elziere wrote:
I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good
stability.
Same here. We've been using ubuntu lucid 32bit for years. We have about
1000 implementations of this.
--
On Wednesday 16 October 2013, Michelle Dupuis wrote:
Is there a recent survey of that Linux distro and version people are using
for the Asterisk installations? I recall seeing a pie chart over a year
ago (I think on a wiki but I can't find it again)also hoping for
something more current.
We are using Debian 32bit and 64bit on standalone and on VMs without any
issue.
On Thu, Oct 17, 2013 at 10:15 AM, Frederic Van Espen
frederic...@gmail.comwrote:
On 10/17/2013 09:47 AM, Alban Elziere wrote:
I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good
stability.
i would vote for debian. simple, STABLE, secure. Most importantly it is
lightweight
On Thu, Oct 17, 2013 at 11:37 AM, Asghar Mohammad asghar...@gmail.comwrote:
We are using Debian 32bit and 64bit on standalone and on VMs without any
issue.
On Thu, Oct 17, 2013 at 10:15 AM, Frederic Van
2013-10-03 09:52, Johan Wilfer skrev:
2013-10-02 17:12, Shaun Ruffell skrev:
On Wed, Oct 02, 2013 at 01:17:15PM +0200, Johan Wilfer wrote:
If I did use the core timers in dahdi (not loading dahdi_dummy) I
got bad quality in the conferences and dahdi_test showed 99.6% as
worst.
Hmm...this is
Is there a recent survey of that Linux distro and version people are
using for the Asterisk installations? I recall seeing a pie chart over a
year ago (I think on a wiki but I can't find it again)also hoping for
something more current.
Mix of Gentoo and Ubuntu here (Gentoo mostly on old
Most tutorials over internet are based on Centos and Ubuntu.
Centos is the base distro of FreePBX, Elastix and Trixbox and always have a lot
of users.
I use ubuntu.
Best regards.
Emiliano
Enviado desde mi BlackBerry de Personal (http://www.personal.com.ar/)
-Original Message-
From:
Hello,
I have a question about best practice (or recommended practice) for allowing
SIP registrations from the Internet.
This is what I was thinking of implementing:
1. Use OpenSips for the SBC, enable SRTP and TLS
2. Allow limited access to the actual Asterisk PBX (behind firewall) via
Le 17/10/2013 12:30, richard.seg...@marisec.ca a écrit :
Hello,
Hello
I have a question about best practice (or recommended practice) for allowing
SIP registrations from the Internet.
Registrations from Internet is vague:
- are EP with fixed IP: define the extension in SIP.conf with
The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under this
scenario. Basically this setup is for people who are traveling, and may be
using a smart phone at an airport (or something similar). The idea is that our
system can be used to reduce toll costs, and provide access
On 13-10-17 08:13 AM, richard.seg...@marisec.ca wrote:
The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under
this scenario. Basically this setup is for people who are traveling, and may
be using a smart phone at an airport (or something similar). The idea is
that our
On Thursday 17 October 2013, richard.seg...@marisec.ca wrote:
The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under
this scenario. Basically this setup is for people who are traveling, and
may be using a smart phone at an airport (or something similar). The idea
is that
If remote users *only* need to call contacts *within the office*, then whatever
other precautions you take, make sure they land in a context which does not
allow outside calls.
Yes, but this is not sufficient. When transfers are allowed, the outside channel will operate in
the local context
Hi,
I try to find some information about CAS E1 signalling and how it's
handled by Asterisk. My customer wants to connect to a BT ITS Netrix by
CAS E1 EM. The system is intended to take the channels and mix them
(meetme / confbridge) and send the audio back mixed to each.
The layout:
BT ITS
Is Asterisk is fully Complaint to RFC 3261 ?
I am facing issue with expire Header , Contact header expire parameter, Max
forward header range.?For e.g RFC 3261 say Max-Forward Header range should be
0-255 Data from RFC-
20.22 Max-Forwards
The Max-Forwards header field must be used with
Hard to give an answer but for us it is
Centos - 1, others - 0
Ron
On 17/10/2013 6:16 AM, emilianovazq...@gmail.com wrote:
Most tutorials over internet are based on Centos and Ubuntu.
Centos is the base distro of FreePBX, Elastix and Trixbox and always have a lot
of users.
I use ubuntu.
Sometimes you do not really have a choice. If you are using TDM cards where you know that the
manufacturers use CentOS for their own development, then it makes no sense to use a different
distro.
Problems you might run into are that init scripts might need a patch here and there, or that
most
On Thu, Oct 17, 2013 at 7:53 AM, Johan Wilfer li...@jttech.se wrote:
1. In asterisk can I get the channel-number of the call so I can have
different logic for the different channels?
Sure, I guess I would just create different incoming contexts for your
various channels in chan_dahdi.conf. Or
On Tue, Oct 15, 2013 at 11:58 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Is there a recent survey of that Linux distro and version people are using
for the Asterisk installations? I recall seeing a pie chart over a year ago
(I think on a wiki but I can't find it again)also hoping for
Hello,
SayNumber() app embeds a smart logic to play numbers according specified
locale. For example, SayNumber(71) would play files sixty, and and
eleven in french.
If I'm not mistaken, if I had to set a dialplan variable with the file
list SayNumber would play for a given number, I would have
Dear list,
on Asterisk 1.4.21 which is being used in a callthrough scenario -
callers call via PSTN to a DID coming in via SIP and then dialing
outbound via DTMF and the outbound calls get routed via some SIP
termination provider - lately I see that every now and then MusicOnHold
gets
Hi,
Is there some SIP magic that can trigger MusicOnHold on my end?
obviously, putting a call on hold will trigger music on hold.
Mayber your gateway does that when all outbound channels are busy or
something?
-nik
--
Wer den Grünkohl nicht ehrt, ist der Mettwurst nicht wert!
Can you post sip debug and the console log for this call?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Thursday, October 17, 2013 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial
Markus wrote:
Started music on hold, class 'default', on
SIP/outbound-sip-provider-0002
I see this on our system and it's considered a feature. When the remote
system signals that a call has been put on hold, it will instruction the
local Asterisk system to do the actual holding.
I see
Enable the full log in logger.conf. then in asterisk CLI run sip set debug on.
Wait for it to happen again, then check the log file /var/log/asterisk/full to
see if there's a sip statement being sent to your server to enable music on hold
Sent from my Verizon Wireless 4G LTE DROID
Dominik
[Apologies, top-posting, Gmail, yadda yadda]
As with a lot of software, I suspect the best answer is whichever distro
YOU are most comfortable with. You're the one who has to support it, after
all... Just my 2c.
Andrew
On Thursday, 17 October 2013, Rusty Newton wrote:
On Tue, Oct 15, 2013 at
debian wheezy compiled asterisk from source
On 18 October 2013 00:27, Andrew Furey andrew.fu...@gmail.com wrote:
[Apologies, top-posting, Gmail, yadda yadda]
As with a lot of software, I suspect the best answer is whichever distro
YOU are most comfortable with. You're the one who has to
Today I was hacked but caught it very quickly. This is the weird part, they
hacked an IP Auth based account by simply knowing the account name.
How is this possible? I am running Asterisk 11.5.0. Now it's my fault I used a
dictionary based account name but how did they bypass the set ip I had
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