You could traceroute the IP and contact NOC's along the route. They might
be interested to hear of flooding/DOS attacks being routed via their
equipment.
On Tue, Mar 6, 2012 at 4:58 PM, Mike Diehl mdi...@diehlnet.com wrote:
route add -host 188.138.100.16 dev lo
Good bye. But it shouldn't
Hi List,
Has anyone heard of an ATA device that supports TCP TLS? Not much
comes up in searching, thought to check here for some device
suggestions.
TIA,
Skyler
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Hi List,
When a user is on a call, sometimes they hear digits dialing as if the
other end is randomly pressing the keypad with their face...but they
aren't. It has happened while I've been on calls also, very odd and
annoying.
Has anyone come across this on Asterisk before?
TIA,
Skyler
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler
Sent: Thursday, December 08, 2011 9:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] random digits dialing during call
Hi List,
When a user
for a project here. Would you mind if I
contacted you off list with some getting started questions?
Skyler
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New to Asterisk? Join us for a live introductory
Hi,
On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote:
Hello,
I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in
sip.conf at
[general] section the following options:
transport=tcp
tcpenable=yes
tcpbindaddr=0.0.0.0
but after all that changes i still not
I have the below in my sip.conf
bindaddr = PublicIP
tcpenable = yes
tcpbindaddr = PublicIP
On Thu, 2011-08-25 at 17:22 +0300, Catalin S. wrote:
hello,
I tried still not working. :( something is wrong.
On Thu, Aug 25, 2011 at 4:37 PM, Skyler skchopper...@gmail.com
wrote
On Wed, 2011-08-24 at 08:49 -0600, Linuxguy123 wrote:
OK. I'm a 54G guy. I just bought a E4200 the other day for our media
network.
Nice. The E4200 is what I wanted but it wasn't in stock anywhere when I
was on the hunt. I just recently bought an Asus RT-N16 ... my boss is
hooked now so I
I have my wireless router working as a proxy/asterisk system. Its not
100% done yet, config related stuff still lingering, works not so bad so
far. I register voip phones or ata's locally and SIP trunk for my
Voxnumber(s), also for inbound/outbound. It does callback call-through
for mobile, also
Hi,
On Tue, 2011-08-23 at 21:50 -0600, Linuxguy123 wrote:
So you have asterisk loaded on a wireless router ? Linksys 54G by
chance ?
Yes, Asterisk at the moment. Cisco E3000. 54G is too small for
asterisk, not enough flash/cpu.
Which VOIP phones are you using ? Which ATA are you using ?
sip.conf
useragent = myasteriskbox
sdpsession = myasteriskbox
On Wed, 2011-07-20 at 12:45 +0300, Israel Gottlieb wrote:
user-agent could be set in sip.conf
On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov
abalas...@evaristesys.com wrote:
On 07/20/2011 05:00 AM, Masood Ahmed wrote:
Hi,
On Tue, 2011-07-19 at 16:14 +0200, Gilles wrote:
On Mon, 18 Jul 2011 20:59:02 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
/usr/lib/asterisk/modules/
Be sure to only include the ones you need. Finding which exactly may be
tricky.
Thanks Tzafrir. Actually, since the modules
Hi,
On Tue, 2011-07-19 at 14:30 -0500, Warren Selby wrote:
On Tue, Jul 19, 2011 at 11:53 AM, motty.cruz motty.c...@gmail.com
wrote:
Hello All,
I have asterisk server running on Centos, some of our users
are spreadout
throut the states. I want the time zone to
Hi,
I had a similar issue converting wav files one time. Ended up using sox to
convert to .sln as that ended up being the sounding conversion.
I used the below command on a directory of files to convert:
for a in *.wav; do sox $a -t raw -r 8000 -s -w -c 1 `echo $a|sed
s/.wav/.sln/` resample
) as the micropayment solution for
user-requested updates. Some nominal fee.
If anyone wants to get involved, contact me.
On 06/01/2011 07:51 AM, Skyler wrote:
Hi,
The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the
carrier's way to isolate their users and another excuse to charge
Hi,
The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the
carrier's way to isolate their users and another excuse to charge more money
for 'the better plan'. In the end, it's the carrier that inputs the info so
if it shows FLORIDA with one database I can't see how any other database
Hi all,
Anyone know if it's possible to force asterisk to use 'my_table' for
passwords instead of the 'secret' table?
S.
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New to Asterisk?
Hi,
You can run the script below as an hourly cron. Works for me.
#!/bin/sh
# clean-up Asterisk zombies
# file clean_up.sh
# $Id: clean_up all dead parent processes
# use as cron task */30 * * * * root /usr/local/sbin/clean_up.sh
#
Hi all,
Anyone know how to make asterisk properly reply to options keep-alive? Or
just force a 200 OK somehow?
I recently took over a server and they have ~80 pap2 devices that send nat
keep-alive and * always replies with 481 No subscription. It's more of an
annoyance, I know but I
Of Skyler
Sent: Friday, May 13, 2011 2:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] OPTIONS Keep alive - Reply: 481 No
subscription
Hi all,
Anyone know how to make asterisk properly reply to options
keep-alive? Or just force a 200 OK
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] concurrent call tracking
On 11-05-11 06:36 PM, Skyler wrote:
Thanks Dovid, if you don't mind sharing the code and the dial plan side
I'd
like to take a look at it for sure. The dial plan example Leif replied
with
is pretty much what
* 1.6.2
+ mysql - realtime (no gui). Any suggestions / open-source / AGI on where to
start looking into implementing something like this?
TIA,
Skyler
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.
I then have a graph that shows channel usage. If you want the code let me
know.
- Original Message -
From: Skyler mailto:skchopper...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2011 19:57
Subject: [asterisk-users] concurrent call tracking
Hi
First, I'm pretty sure avaya peer needs to friend. Try adding the below to
sip.conf and do a reload.
[general]
externip = the.wan.ext.ip
localnet = 192.168.1.0/255.255.255.0
If that doesn't work, add nat=yes to avaya peer=friend
Skyler
From: asterisk-users-boun
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