search for: 0123456

Displaying 15 results from an estimated 15 matches for "0123456".

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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