?
--
Regards,
Giles Coochey, CCNP, CCNA, CCNAS
NetSecSpec Ltd
+44 (0) 8444 780677
+44 (0) 7584 634135
http://www.coochey.net
http://www.netsecspec.co.uk
gi...@coochey.net
smime.p7s
Description: S/MIME Cryptographic Signature
On 08/06/2014 22:01, Mark Robinson wrote:
Hello,
can someone recommend a good and free Softphone for Windows which does
not display advertisments inside the program?
Has anyone tried MicroSIP?
http://www.microsip.org/
--
Regards,
Giles Coochey, CCNP, CCNA, CCNAS
NetSecSpec Ltd
+44 (0
that way for many many years.
With Asterisk 11 (at least), if you use the security log, it does show
the source IP.
I have configured fail2ban to automatically block these, as shown here:
http://www.coochey.net/?p=61
--
Regards,
Giles Coochey, CCNP, CCNA, CCNAS
NetSecSpec Ltd
+44 (0) 8444 780677
be wanting to set up covert,
untrackable communications channels, but unlikely in my opinion.
--
Regards,
Giles Coochey, CCNP, CCNA, CCNAS
NetSecSpec Ltd
+44 (0) 8444 780677
+44 (0) 7983 877438
http://www.coochey.net
http://www.netsecspec.co.uk
gi...@coochey.net
smime.p7s
Description: S/MIME
Asterisk which FreePBX were not able to stop before.
I would welcome any improvements anyone would care to submit and I'll
extend the article a little.
The changes need the Asterisk security log feature, which I think was
only introduced in later versions of Asterisk (e.g. v11).
--
Regards,
Giles
On 08/07/2013 16:11, Patrick Lists wrote:
On 07/08/2013 01:46 PM, Giles Coochey wrote:
Just a note that I did a little work to extend FreePBX distro with some
extra Fail2Ban which deals with some drive-by SIP registration attempts.
My regex is poor to middling, but the steps detailed here
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Title: Re: [Asterisk-Users] TDM01B vs. X100P
Hi Chris,
I have this more or less working, I can dial the IP Office
extensions directly from Asterisk.
How do I configure being able to dial Asterisk VoIP
extensions directly from IP Office phones??
Currently I have a short code to dial
Title: Newbie Question: Help with incoming dial plan
exten =
s,1,Answerexten = s,2,Wait,2exten =
s,3,Background(enter-ext-of-person)exten = s,4,DigitTimeout,5exten
= s,5,ResponseTimeout,10
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: 18
Hi All,
I am trying to connect to cvs.digium.com but connection gets
timed out.
Even pinging to cvs.digium.com is not working.
I m using
cvs login
password - anoncvs
I'm able to checkout the Asterisk module right now using the following
CVSROOT:
:pserver:anoncvs:[EMAIL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adrien Laurent
Sent: 02 August 2005 14:56
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to create a secret code to use
[EMAIL PROTECTED] server's long distance plan from a
FYI
If using oh323 v0.6.5 then the oh323 show info has been replaced by
the command oh323 show channels
Thanks
Giles
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: 31 May 2005 12:51
To: asterisk-users@lists.digium.com
Subject:
Has anyone seen a situation where, upon connecting two
asterisk servers
together with IAX registration, outgoing/incoming calls that
route through
both servers are choppy and jittery? I don't have this
problem when I call
out to teliax (my ITSP) directly, but if I try to make the
Title: Message
Google
is your friend:
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2003-36,GGLD:enq=%22What+is+Asterisk%40Home%22
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
QuintinSent: 24 May 2005 11:41To:
How do you disable hyper threading (what's the command and where is it
placed)?
Hyper-threading is a BIOS feature available on some Pentium 4 Xeon
processors. If you have hyper-threading enabled your system may appear
to have more processors than are physically in the system. Typically
Moreover, The FS108P can only power 4 ports simultaneously.
I'd prefer something like this:
http://www.netgear.com/products/details/FSM7326P.php
Or a Cisco equivalent.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Mason (Lists)
Sent: 16
I had to use google translate to answer your question, so I'm going to reply
with my answer in the same way, the extra time it took me to decipher the
question will probably have to be reinvested:
Le safe_asterisk est juste un manuscrit pour remettre en marche le serveur si
il se termine
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Mason (Lists)
Sent: 19 April 2005 13:57
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] VPN/Asterisk combo
How do you get 18 interfaces on one
Probably VLANs and a router before the firewall.
If you use VLANs, woud all the computers be able to access
all the resoures
on the network?
Only if you route between the VLANs. You can enable dot1q tagging in
Linux and sub-interface the PBX, then configure that port as a dot1q
VLAN
Jalal wrote:
no firefox no linux no asterisk .
You guys should check really before posting, this link works fine for me
in both Konqueror and Firefox, and additionally:
http://uptime.netcraft.com/up/graph/?host=www.callaccounting.ws
The site appears to be running Linux.
Who cares?
Bye .
Greetings!
This is my first post to the list...and I'm kinda' new to Asterisk, so
please be kindI did a fair amount of Googling but was not able to
find an answer.
I am using [EMAIL PROTECTED] 0.8
I was wondering if there is a way to select the outbound trunk based
on the
Hi,
I know this can be done but I guess I am not understanding
the few notes
I have seen on this - can SIP phones be tied to Asterisk
using a PC mac
address instead of their IP address (obviously I am using DHCP). If
someone could please point to the right Wiki or How to I
would
Hi,
I am new to Asterisk and the first thing I have noticed about
Asterisk
and Pingtels open PBX's is that they are using this dinosaur
method of
running forums. It is a real pain getting every message in
the forum and
essentially keeping my own database of issues. With that said
Ok, I am still working on getting this PolyCom phone working
with Asterisk.
I have been looking all over, but I have not been able to
find the username
and password for the web interface on this phone.
I found some site that said it was Polycom and spip, but that
does not work.
I get this error:
if [ -d CVS ] ! [ -f .version ]; then echo
CVS-v1-0-03/29/05-15:19:53
.version; fi
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o
config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o
ulaw.o alaw.o callerid.o fskmodem.o image.o
How about scanning for it's mac address?
http://ipscan.sf.net/ipscan.exe
--
http://www.umich2.com
Digium doesn't label the MAC address on the device, unless it's such a
fine print that no one can read it. I believe this has been said a few
times in the conversation.
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