Hi,
I have 2 port Digium PRI card (TE205P). On each port we have different
service provider E1 is connected.
I have to configure if i dial with prefix 910 it has to dial out
through port 1 service provider and if i dial prefix 912 it has to
dial out through port 2.
My zapatel.conf:
On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote:
Perhaps if there was a Asterisk RBL we could all contribute to; for which we
could then hook into and drop any connection where a source IP is listed ?
--
Thanks, Phil
I love the idea of a RBL... count me in for contributing.
- Original Message -
Am 11.04.2010 17:05, schrieb Mark Smith:
Same this end from 184.73.17.150.
Use this little piece of iptables magic to block the whole of
Amazon's EC2 ip-
range.
iptables -F
iptables -A INPUT -m iprange --src-range
216.182.224.0-216.182.239.255 -j DROP
I got the same generic response, asking me to submit the same info which I
had already submitted. This clearly show they are not interested in tracing
just another hacker on their cloud.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-12 9:24 AM, Fred Posner
If RBL or something is practical, I'm in too. But at what level these
hackers will be blocked? Unless some big ISPs cooprate, it is not much of
use.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-12 9:24 AM, Fred Posner f...@teamforrest.com wrote:
On Apr 12, 2010,
On Apr 12, 2010, at 8:17 AM, Fred Posner wrote:
On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote:
Perhaps if there was a Asterisk RBL we could all contribute to; for which we
could then hook into and drop any connection where a source IP is listed ?
--
Thanks, Phil
I love the
On Mon, Apr 12, 2010 at 3:52 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
If RBL or something is practical, I'm in too. But at what level these
hackers will be blocked? Unless some big ISPs cooprate, it is not much of
use.
I've been following this with much interest. I don't see RBL (which I
This thread needs to go into a RBL - guess I'm being part of the problem,
not the solution...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Owen
Sent: Monday, April 12, 2010 9:04 AM
To: Asterisk Users
Michael Nausch wrote:
HI,
I tried to install asterisk and mISDN via
http://www.asterisk.org/downloads/yum
My machine is running with kernel-2.6.18-164.15.1.el5.i686
Packages for that kernel version were missing. That was an oversight and has
been corrected. A `yum update` should be
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a
way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
make menuselect.makeopts
Olivier wrote:
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such
a way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
2010/4/12 Olivier oza_4...@yahoo.fr
Hi,
In my 1.6.1.18 with dahdi 2.2.1.1, I've got :
# dahdi_hardware
pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card
Does it mean I should download and use qozap or is it a bug in Dahdi ?
Regards
I should have added that I'm
Hi,
In my 1.6.1.18 with dahdi 2.2.1.1, I've got :
# dahdi_hardware
pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card
Does it mean I should download and use qozap or is it a bug in Dahdi ?
Regards
--
_
--
Good article - might solve our problems for now:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood
He got the bots to stop by writing a ruby script that responds back to them
with a SIP 200 OK.
I'm going give it a go when I'm back home...
Cheers,
Tom
--
On 12 Apr 2010, at 17:30, Tom Stordy-Allison wrote:
Good article - might solve our problems for now:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood
He got the bots to stop by writing a ruby script that responds back to them
with a SIP 200 OK.
I'm going give it a
On 04/12/2010 08:17 AM, Fred Posner wrote:
On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote:
Perhaps if there was a Asterisk RBL we could all contribute to; for
which we could then hook into and drop any connection where a
source IP is listed ? -- Thanks, Phil
I love the idea of a
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project. It's a
blacklist of IP addresses and address ranges which are known to ONLY be
used for malicious purposes.
http://www.spamhaus.org/drop/
On Apr 12, 2010, at 1:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project. It's a
blacklist of IP addresses and address ranges which are known to ONLY be
used for malicious
Hi Alyed,
Thank you for the response. I tried this solution, I got Unknown
displayed instead of 999. Also, I tried both 200 and 200 as the CID
number for the extension, but the results were the same.
On Sat, Apr 10, 2010 at 2:10 PM, Alyed al...@vivoxie.com wrote:
Don't have a system to test
On 04/12/2010 12:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project. It's a
blacklist of IP addresses and address ranges which are known to ONLY be
used for malicious
On Sun, Apr 11, 2010 at 5:00 PM, Danny Dias ing.diasda...@gmail.com wrote:
This digium card has 3 FXO ports and 1 FXS port where we have a fax
machine
connected!
The problem is that we can receive fax very good, but we can't make any
outbound fax call, in fact, our asterisk get freezed in
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
A phone with a good speaker phone.
Ability to make the audio one way. I want to monitor the room but not
have my voice heard in the room. Yes, the mute button can
- Original Message -
On 04/12/2010 12:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project.
It's a
blacklist of IP addresses and address ranges which are known to
HI Jason!
Am Montag, den 12.04.2010, 10:39 -0500 schrieb Jason Parker:
Packages for that kernel version were missing.
Jepp, I thought so.
That was an oversight and has been corrected.
No problem, verybody can make an error, me too, :)
A `yum update` should be enough to solve this for
Hi all,
I'm trying to configure an * box for my home in an embedded device, so
I want a minimum configuration. I've already configured it to connect
to my SIP provider and my IP phones and ATAs, so far so good. My SIP
provider gives me voicemail service and I'm happy with it. I don't
want to run
Hi,
What can I make of the following log messages? Extension 7114 tries to reach
6035 but gets an unknown channel type. What does it mean? (supposedly, 6035
was not busy...)
Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing
Dial(SIP/7114-b4fe1ef0, /6035|300|) in new stack
Apr 12
- Vieri rentor...@yahoo.com wrote:
Hi,
What can I make of the following log messages? Extension 7114 tries to
reach 6035 but gets an unknown channel type. What does it mean?
(supposedly, 6035 was not busy...)
Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing
I know that Asterisk can use the system's sound card as the output device
for a console channel. However, I'm using Asterisk call files and would
like to be able to hear the calls over a set of speakers as the call files
are being processed. Basically I'm wanting to listen in on the calls as
Somebody posted a thread last week about redirecting a channel to XML using
EAGI - that's the direction you probably want to go.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Monday, April 12, 2010 1:40
On Fri, Apr 9, 2010 at 5:40 PM, Joe Freeman j...@ngn-networks.com wrote:
I have res_fax setup and working for the most part. However, I'm seeing
some fax machines drop the connection on me -
Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel
'DAHDI/1-1' did not return a
On Fri, Apr 9, 2010 at 3:26 PM, Aldo Bergamini aabe...@nb-a.com wrote:
Hi all,
I have noticed something I can't solve regarding Asterisk (latest
1.6.0.x).
My server is set at the GMT+2 timezone. The clock is ok (I can get the
correct time at the terminal). But today I got a call at a time
Darrick Hartman wrote:
On 04/12/2010 12:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
snip /
Randy,
That only addresses EC2 (and assumes that Amazon has any interest in
protecting their reputation). What about attacks that come
On Mon, Apr 12, 2010 at 2:23 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Vieri rentor...@yahoo.com wrote:
Hi,
What can I make of the following log messages? Extension 7114 tries to
reach 6035 but gets an unknown channel type. What does it mean?
(supposedly, 6035 was not busy...)
Apr
Hi Guys,
Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2,
LibPRI 1.4.10.
Placing a call into PRI and then transfering that call out to another
number. Problem is that the call rings out but the moment the other party
pickups both legs of the call are disconnected give
HI Jason,
Am Montag, den 12.04.2010, 10:39 -0500 schrieb Jason Parker:
That was an oversight and has been corrected.
I'm not sure If I on the right place, but I've a little improvment
suggestion.
The maintainer of that mISDN RPM may append the startupscript at the
beginning with:
- bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI
1.4.10.
...etc
I was going to respond with some very insightful and helpful information but
I'm not a PRI Guru. Sorry, maybe next time.
--Tim
--
It is normal for the PSTN switch to disconnect both channels when a Two
B-Channel Transfer is completed successfully.
Are the two parties connected?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From:
I'm currently receiving over 200 SIP REGISTER requests per second from a
machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it.
This has continued for several days, and ab...@staff.aruba.it are
unresponsive. I've had a couple of similar incidents recently, the
others originating
On Apr 12, 2010, at 4:50 PM, Chris Hastie wrote:
I'm currently receiving over 200 SIP REGISTER requests per second from a
machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it.
This has continued for several days, and ab...@staff.aruba.it are
unresponsive. I've had a couple
Perhaps if there was a Asterisk RBL we could all contribute to; for
which we could then hook into and drop any connection where a
source IP is listed ? -- Thanks, Phil
I love the idea of a RBL... count me in for contributing.
Especially considering the ridiculous response I received from
Please shed some lights if you can see the source of the problem in the
debug. The subject was not meant to be a deterrent but rather emphasizing
the complexity of issue at hand. As I noted at the bottom of my post, I
appreciate any and all input.
-Bruce
On Mon, Apr 12, 2010 at 4:02 PM, Tim
Thanks for the input Don.
HmmmI am not understanding the comment here. I am not doing any flash()
or transfer() but rather just dial out and native zap bridge should just
connect two channels and only hangup both channel when one party hangs up.
Here is what should happen:
Call comes in and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner
Sent: 12 April 2010 21:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Flood of REGISTERs - attack?
On
On 12 Apr 2010, at 20:00, David Backeberg wrote:
chan_carrierpigeon must be in asterisk-extras. I'll have to upgrade
and check it out.
Terrible latency, and seems susceptible to packet loss where shotguns are
involved.
S
--
Viking electronics analog phones (I think E20) connected to an ATA.
Or cyberdata door boxes.
On Mon, Apr 12, 2010 at 1:19 PM, Mark Hulber asterisk.ad...@hulber.com wrote:
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
It just hit me that you are talking about TBCT. I don't think I am doing
TBCT as I still want both channels to keep two lines of my PRI occupied. In
addition, I would be interested to know how TBCT is done over PRI. I know
that this can be done over analogue with flash().
Can you please elaborate
Hi Guys,
I am sorry if my issue is not related to this but I think it is.
I have a PRI with Bell Canada and when I dial in and have the call
transfered to a context to dial out and then have those two channels
bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth
shows in
Futher check into the PRI debug I am seeing this which actually relates to
TBCT and AOC-E error in /usr/src/libpri/pri_facility.c:
Message type: FACILITY (98)
[1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 03]
Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06,
Told you it was too complex of an issue :-) I figured to do this in
zapata.conf and all is fine now:
transfer=no
That was the magic two letter which was sending a request for RLT feature on
the line. Set transfer to no and all worries gone.
Thanks for the input everyone.
-Bruce
On Mon, Apr 12,
Problem resolved with setting transfer=no in zapata.conf.
On Mon, Apr 12, 2010 at 9:14 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
I am sorry if my issue is not related to this but I think it is.
I have a PRI with Bell Canada and when I dial in and have the call
transfered to a
The symptoms look like you're doing TBCT. Unless you're recording or, for
some other reason, want to supervise the call, TBCT is a more efficient use
of your PRI as it frees up channels after the transfer. TBCT isn't available
with analog circuits, but is very similar to the analog flash and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred
Posner
Sent: 12 April 2010 21:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Flood of REGISTERs -
Hello,
im having trouble with the following:
[Asterisk]--[ISP]--[ADSL Modem]--[Linksys
Router]--[Grandstream ATA]--[Analog Phone]
On server:
- Asterisk 1.6
- A2Billing 1.4
A2Billing have 2 Trunks:
- TrExt: Voip Provider
- TrInt: Internal Calls
This structure works on
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