Displaying 15 results from an estimated 15 matches for "0123456".
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123456
2018 Nov 22
3
OpenSSL 1.1.0j version checking bug in configure.ac [PATCH]
...figure.ac
index 7379ab35..dcf52230 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2616,7 +2616,7 @@ if test "x$openssl" = "xyes" ; then
AC_MSG_ERROR([OpenSSL >= 1.0.1 required (have "$ssl_library_ver")])
;;
100*) ;; # 1.0.x
- 101000[0123456]*)
+ 101000@<:@0123456@:>@*)
# https://github.com/openssl/openssl/pull/4613
AC_MSG_ERROR([OpenSSL 1.1.x versions prior to 1.1.0g have a bug that breaks their use with OpenSSH (have "$ssl_library_ver")])
;;
2015 Jun 08
6
Am I cracked?
Hi list!
Very strange...
I ran the Asterisk CLI for other tasks, and suddenly I got this message:
== Using SIP RTP CoS mark 5
-- Executing [000972592603325 at default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack
== PROXY Call from 0123456 to 000972592603325
-- Executing [000972592603325 at default:2] Set("SIP/192.168.20.120-0000002a", "CHANNEL(musicclass)=default") in new stack
-- Executing [000972592603325 at default:3] GotoIf("SIP/...
2004 Dec 31
2
CUPS/NFS
Is there anyway to remove CUPS and NFS on the system, I'd like to stop these
services, close the ports, and uninstall the software as I don't use NFS for
anything on my system, and for sure don't use printing on the server.
Regards,
Joshua Strzalko
President, Eleet Technologies, Inc.
Office: 407.977.2857
Mobile: 407.923.6926
Fax: 321.244.9435
josh at eleet-tech.com
2011 Jun 20
1
Problems with pickupgroup/callgroup with Asterisk 1.8.4.2
...er (pid = 3915)
Verbosity was 3 and is now 6
-- Accepting AUTHENTICATED call from 111.222.333.444
:
> requested format = alaw,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (ulaw|alaw),
> priority = mine
-- Executing [0123456 at from_trunks:1]
SIPAddHeader("IAX2/khca_in-2443", "Alert-I
nfo: <http://127.0.0.1>\;info=external") in new stack
-- Executing [0123456 at from_trunks:2] Goto("IAX2/khca_in-2443",
-- Executing [0123456 at from_trunks:3] Dial(&quo...
2007 Feb 27
0
asterisk CDR and mysql
Hello all,
I added a record named pre_dst in the cdr table.
It has the same type as dst field.
And I used this line in the dialplan:
exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
When I call, 70123456, (7 is only to use the provider trunk),
I have this in the CLI:
Executing Set("SIP/foo-0816a490", "CDR(predst)=0123456") in new stack
-- Executing NoOp("SIP/foo-0816a490", "0123456") in new stack
-- Executing Dial("SIP/moi-0816a490", "SIP/01234...
2015 Jun 08
0
Am I cracked?
...I suspect this is middle east), look for suspicious dialing patterns, etc.
If you still have allow guest enabled, then you should also follow the 'securing asterisk' steps from this site: http://www.voip-info.org/wiki/view/Asterisk+security
You're definitely under attack (based on the 0123456 ID) so be sure to take preventative steps to avoid a $50k phone bill..
________________________________________
From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Luca Bertoncello <lucabert at lucabert.de>
Sent: Monday, June 8, 20...
2005 Aug 08
3
Digium TE405P, caller id and migration to *
Hi,
we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our
old PBX. So now we could migrate to the * server.
But, there are two things we can't live with:
1. A call from the outside to the old PBX is missing a leading 0 before the number.
Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as
caller number.
2. A call made from a SIP client to the outside lacks the extension in the number:
Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number
like 6789-234 when dialing out over the PSTN.
Can anybody tell me...
2004 Sep 11
2
Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do?
...he Wiki and Googled the hours away, but I still can't
find supportive documentation.
I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap
configuration and had the following problems:
1 - Outbound dialed numbers had the MSD stripped off
So, I issue the command 'Dial(Zap/g1/0123456) and Zap tries to dial
'123456'. It took a bit of digging to actually work out what the problem was
in the first place, but I managed to resolve this by setting
'pridialplan=unknown' in zapata.conf.
2 - Inbound CallerIDs have the MSD stripped off (only tested with UK
numbers, where...
2015 Jun 08
0
Am I cracked?
> Very strange...
> I ran the Asterisk CLI for other tasks, and suddenly I got this message:
>
> == Using SIP RTP CoS mark 5
> -- Executing [000972592603325 at default:1] Verbose("SIP/192.168.
> 20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") innew
stack
> == PROXY Call from 0123456 to 000972592603325
> -- Executing [000972592603325 at default:2] Set("SIP/192.168.20.
> 120-0000002a", "CHANNEL(musicclass)=default") in new stack
> -- Executing [000972592603325 at default...
2015 Jun 10
0
Am I cracked?
...t; wrote:
> Hi list!
>
> Very strange...
> I ran the Asterisk CLI for other tasks, and suddenly I got this message:
>
> == Using SIP RTP CoS mark 5
> -- Executing [000972592603325 at default:1]
> Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to
> 000972592603325") in new stack
> == PROXY Call from 0123456 to 000972592603325
> -- Executing [000972592603325 at default:2]
> Set("SIP/192.168.20.120-0000002a", "CHANNEL(musicclass)=default") in new
> stack
> -- Executing [000972592603325 a...
2014 Sep 19
0
vsnprintf_s and vsnprintf
...use vnsprintf?
>
> Ok, we need to drop vsnprintf_s to support WinXP. I'd prefer vsnprintf
> over vsprintf but have no way of testing any of these options.
I wrote a small program that fills a buffer[] with "abcdefghijklmnopqrstuvwxyz\0"
pattern and then tries to write "0123456789" string into it.
It calls either
ret = vsnprintf_s(buffer, buf_size, _TRUNCATE, fmt, va);
or
ret = vsnprintf(buffer, buf_size, fmt, va);
The results are:
---------------------------------------------------------------------------
vsnprintf_s (MSVC, MinGW):
buf_size = 8; r...
2004 Jul 13
1
HFC-S card and Unable to create channel of type 'Zap'
...A1
hi,
i'm new to *
I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2;
when i try to call outside i get:
-- Accepting AUTHENTICATED call from 192.168.1.110, requested format = 1024, actual format = 1024
-- Executing Dial("IAX2[pippo@pippo]/2", "Zap/g1/0123456") in new stack
Jul 13 13:42:49 NOTICE[884752]: app_dial.c:559 dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy at this time
Jul 13 13:43:07 WARNING[163851]: chan_zap.c:6070 zt_pri_error: PRI: Read on 19 failed: Unknown error 500
Jul 13 13:43:07 NOTICE[163851]: ch...
2014 Sep 18
3
patch for win_utf8_io.c: vsnprintf_s vs. MinGW
lvqcl wrote:
> Oops. It seems that vsnprintf_s isn't always available on MinGW platform:
> MinGW declares this function only if MINGW_HAS_SECURE_API macro is defined.
> That's because WinXP version of msvcrt.dll doesn't contain secure functions
> like vsnprintf_s.
>
> Maybe it's better to revert vsnprintf_s to vsprintf or to use vnsprintf?
Ok, we need to drop
2006 Jan 27
1
chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1
...how to tell if it should work or not later).
> > > > >
> > > > > Here's the relevant instructions on
> > > > > http://www.crazygreek.co.uk/content/chan_bluetooth for how to dial
> > > > > out: Send a call out by using Dial(BLT/DevName/0123456).
> > > > >
> > > > > As far as dialing in, there's a special context (I think [bluetooth]
> > > > > maybe? I'll have to get back to you on that). I know that it should
> > > > > work fine, because I tried dialing the phone, aste...
2009 Nov 10
23
Puppet Not Running chkconfig correctly.
So, I noticed that when adding a service to puppet, puppet is running
a chkconfig <service>. However, as far as I can tell, puppet should be
running a chkconfig --add <service>, otherwise the symlinks don''t get
created from the service in /etc/rc.d/rc6.d back to the service in
/etc/init.d. As as a result, the service is not stopped on shut down.
I must be missing something