On Mon, 28 Feb 2011 12:12:44 +1000, Stuart Longland
redhat...@gentoo.org wrote:
Apart from the lack of any hardware signal processing, it seems all the
components are there. The server isn't particularly heavily loaded, and
thus I see no reason why the machine wouldn't theoretically be able to
I use Yealink phones too.
These phones are quite good, recommend them.
Com os melhores cumprimentos,
Best regards,
CÉSAR SEQUEIRA
IT Expert
M: +351 961 355 772
@: cesar-seque...@justbit.pt
skype: cesar.sequeira.justbit
msn: cesar-seque...@justbit.pt
-Mensagem original-
De:
Hi All;
I would like to have two Asterisk machines to have redundancy between them, so
if first machine failed then we can depend on the second machine.
Because of this, I would like to know (if someone can advise me):
1) If I did modification on the configuration, how this will be applied to
Hello
Surprisingly, Google didn't return any thread in this ng about the
Obi110 device, which is a new alternative to the 3102:
www.nerdvittles.com/?p=720
I'd like some feedback from Asterisk users who have tried the Obi110
to connect it to a landline.
Thank you.
--
Hi all,
The problem I have been experiencing since last month is that some of my
customers are getting calls with Asterisk Unknown caller id. Most of
them in the middle of the night. And my asterisk server has no record of
these calls. The customers were getting irritated as you can imagine. I
hi
using database as realtime functions solves your first problem,
for second try by using dns
best
On Mon, Feb 28, 2011 at 1:54 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I would like to have two Asterisk machines to have redundancy between them,
so if first machine failed then we
On 28 Feb 2011, at 10:33, Rizwan Hisham wrote:
The problem I have been experiencing since last month is that some of my
customers are getting calls with Asterisk Unknown caller id. Most of them
in the middle of the night. And my asterisk server has no record of these
calls. The customers
On Monday 28 Feb 2011, Steven Howes wrote:
'asterisk security' is a misleading subject line. Guessing someone just
scanned some IP addresses and made calls. You need what's called a
'firewall'.
Well, assuming you're on Linux then you've already *got* a firewall. Just add
some iptables rules
Probably, you are receiving INVITE attacks from some tool like sipvicious.
You should rearange your network to cover some inportant security issues.
The IP address of you server can be revealed in some unincrypted SIP
signaling of some call through the Internet to/from your server's client, or
On 02/28/11 12:36, asterisk asterisk wrote:
I wonder why you do not have the built in ethernet in your motherboard.
Indeed, most motherboards do come with Ethernet on board. This one came
with one gigabit Ethernet interface. However, we needed another for a
connection to an ADSL router (acting
On Sun, 27 Feb 2011 13:32:13 +0100, Gilles codecompl...@free.fr
wrote:
After editing the DAHDI_DEFAULT_FLASHTIME accordingly, I
recompiled/upgraded Dahdi, and ran the script, but Flash() still hangs
up the call:
Turns out that instead or in addition to the above, SHORT_FLASH_TIME
should be
thanks for the replies.
I dont want to rule-out the possibility of network sniffing. I am sure its
not an inside job. The server is off-site and is hosted by a very well
reputed hosting company. So if someone is sniffing, what should I do?
Probably, you are receiving INVITE attacks from some
When he says customers I am assuming he means remote customers. It
sounds like he is a reseller of telecom facilities to me. Which means
his customers most likely have ATA's with port 5060 forwarded to the
ATA, or they are direct on the I'net.
He has already set the ATA to only allow calls from
Database: but this require to use Asterisk version that support Database,
correct? Which Asterisk version does this?
About DNS, please note I am talking about E1 connected to the server, so I was
mean, in case first Asterisk down, then to send E1 calls to second Asterisk. I
am not talking
On Mon, Feb 28, 2011 at 02:24:23AM -0800, bilal ghayyad wrote:
1) If I did modification on the configuration, how this will be
applied to the other machine?
My solution was to create a rsync script to copy the configuration of
the primary server to the standby (every 15m).
2) I am going to
You are right Terry. Sorry i did not describe full scenario before. Yes the
users are remote with atas on port 5060. Attacks on the remote customers was
my second guess. My network/system admin has already ruled out the
implementation of VPN. In summary, we dont want to do anything on remote
On 02/28/2011 10:12 AM, Stuart Longland wrote:
Hi all,
I've tried researching this, and so far, have struggled to find any
contemporary information on the issue, so I do apologise if asking this
irritates people who have answered this before.
I have managed to set up Asterisk 1.8 on the web
Any suggestions on encrypting the sip and rtp. I have done some googling on
it. looks like it is not supported by most end point devices or service
providers. But still your thoughts will be appreciated on this subject.
On Mon, Feb 28, 2011 at 6:13 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Rsync to sync /etc/asterisk and use keepalived/heartbeat for a failover
Asterisk IP.
Make sure to read this -
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+SolutionsFor
From IP rewrite
On Mon, Feb 28, 2011
On 02/28/2011 07:27 AM, Rizwan Hisham wrote:
Any suggestions on encrypting the sip and rtp. I have done some googling
on it. looks like it is not supported by most end point devices or
service providers. But still your thoughts will be appreciated on this
subject.
You cannot protect a remote
I've just installed 1.8.3-rc3 on a test server as we really needed that
deadlock involving REFER fix on our server but now I'm having an odd
issue with one way audio with a specific type of call.
If I do extension to extension calls there is full 2 way audio.
If I route in an incoming call
Hi
try this pls
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18671
it did help to me
On Thu, Feb 24, 2011 at 9:31 PM, Rodrigo Lang rodrigoferreiral...@gmail.com
wrote:
Hi to all!
I'm trying to create a context for integration with extensions.lua and
libsql.mysql, but I'm
Thanks a lot!
Best regards,
Rodrigo Lang.
2011/2/28 Borin katerin.bo...@gmail.com
Hi
try this pls
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18671
it did help to me
On Thu, Feb 24, 2011 at 9:31 PM, Rodrigo Lang
rodrigoferreiral...@gmail.com wrote:
Hi to all!
I'm
Thanks Mr. Kevin.
Can anyone please also tell me which firewall is best suited for
asterisk/sip attack prevention. Is there any firewall built specially to
address sip security problems?
On Mon, Feb 28, 2011 at 6:38 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/28/2011 07:27 AM, Rizwan
http://sipera.com/ is one such product.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Monday, February 28, 2011 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hi,
I am doing failover routing based on 2 dial commands. First route sends back
4xx response and I don't want it to try 2nd route when it is 4xx response.
Can we do failover routing based on SIP 5xx response only ?
Thanks
Deepika
--
Hi,
Would someone know where I can download the CEL schema for (create commands)
for PostgreSQL please ?
--
Thanks, Phil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
The Asterisk Development Team has announced the release of Asterisk 1.4.40. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.40 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 1.6.2.17.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.17 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 1.8.3. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3 resolves several issues reported by the community
and would have not been possible
It could be possible they are not scanning your asterisk server. They are just
scanning 5060 and in this case your ATA caught by scan directly that why you
don't have any logs on server side. Don't you have any setting in ATA to
specify allowed IP address ?
-Satish
From:
Pretty sure I saw those on wiki.asterisk.org.
Thanks,
--Warren Selby, dCAP
On Feb 28, 2011, at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
Would someone know where I can download the CEL schema for (create commands)
for PostgreSQL please ?
--
Thanks, Phil
--
- Original Message -
Pretty sure I saw those on wiki.asterisk.org .
Thanks,
--Warren Selby, dCAP
On Feb 28, 2011, at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Hi,
Would someone know where I can download the CEL schema for (create
commands) for PostgreSQL please
using realtime functions has been added since 1.2.x,
about E1, I thought you are using external voip gateway like cisco,... if
you are using e1 as dahdi driver you have not redundancy option(as I think)
On Mon, Feb 28, 2011 at 4:21 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Database: but
You might want to look at a product called astribank made by Xorcom, which
does a pretty good job of giving redundancy on E1. In their redundant asterisk
setups, they use drbd to mirror configuration and data between asterisk hosts.
Aaron
From: asterisk-users-boun...@lists.digium.com
Google for the DRBD and Heartbeat solution.
you can justget DRBD take care of replicating the information and Heartbeat
to monitoring the primary Asterisk.
On Mon, Feb 28, 2011 at 5:47 PM, Aaron Roberts arobe...@domicilium.comwrote:
You might want to look at a product called “astribank”
Hi again,
since nobody replied yet, maybe I really am the only one experiencing this? Or
is this normal in a way, that I'm stupid to even ask?
now, a day later, it's looking like this:
asterisk*CLI sip show channels
Peer User/ANR Call ID Format Hold
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johannes Jakob
Sent: Monday, February 28, 2011 3:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 1.8.2.4: SIP dialogs not killed?
Hi
Hi,
I'm wondering if this is normal asterisk behaviour:
asterisk*CLI sip show channels
Peer User/ANR Call ID Format Hold
Last MessageExpiry Peer
10.12.0.2(None) 3c2f7ff2975e-wp 0x0 (nothing)No Rx:
PUBLISH
Look like you should work with channel status variable. If channel not
answer then jump on 5xx
--
Sent from my iPhone
On Feb 28, 2011, at 10:27 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com
wrote:
Hi,
I am doing failover routing based on 2 dial commands. First route
sends back
As far I know asterisk doesn't handle the publish sip dialog so it just keeps
it hanging around in 1.8.X (in previous versions it didn't)
I turned off all publish dialogs in the snom phones I have and that got rid of
that
It doesn't really have any impact on the system as far as I have seen
On Feb 28, 2011, at 5:02 PM, isr...@gmail.com wrote:
As far I know asterisk doesn't handle the publish sip dialog so it just keeps
it hanging around in 1.8.X (in previous versions it didn't)
Asterisk 1.8 does handle PUBLISH dialogs, which is why they stay around.
--
Hi,
On 28.02.2011, at 22:55, Danny Nicholas wrote:
If it is affecting your
system performance, post again (and try to use your nice voice :) ).
sorry, I really didn't want to be unfriendly. English isn't my mother tongue,
so I might get the wrong tone sometimes... sorry for that.
On
I'm in the process of testing a TLS/SRTP install. My experience is
improving with each new challenge, but this one is a great test of my 2
month experience with Asterisk.
When I dial 6003 from 6001, it takes 35 seconds until I get the error
message that 6003 is circuit-busy.
Any help would
I just updated from zaptel to dahdi 2.4.0
I dont recall missing keys or duplicating keys with zaptel.
With dahdi 2.4.0 I tried serveral calls and I was trying to enter 204
and I got 2204.
Is there another or new parameter I need to tweek?
Thanks,
jerry
--
2011/2/28 César Sequeira cesar-seque...@justbit.pt
I use Yealink phones too.
These phones are quite good, recommend them.
Com os melhores cumprimentos,
Best regards,
CÉSAR SEQUEIRA
IT Expert
M: +351 961 355 772
@: cesar-seque...@justbit.pt
skype: cesar.sequeira.justbit
msn:
Hopefully this is a simple question.
How does a non-secure phone that is on a PBX connected to an asterisk over a
SIP trunk communicate with a secure phone connected to the Asterisk server?
I like to think that the secure call terminates on the Asterisk and the
non-secure call is somehow
Jerry Geis wrote:
I just updated from zaptel to dahdi 2.4.0
I dont recall missing keys or duplicating keys with zaptel.
With dahdi 2.4.0 I tried serveral calls and I was trying to enter 204
and I got 2204.
Is there another or new parameter I need to tweek?
Thanks,
jerry
I stumbled upon
On Mon, Feb 28, 2011 at 8:49 PM, Mitch Johnson mitch.johns...@gmail.comwrote:
How does a non-secure phone that is on a PBX connected to an asterisk over
a SIP trunk communicate with a secure phone connected to the Asterisk
server?
I think, although I'm not positive, that if either leg of the
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