search for: tih

On Fri, Nov 2, 2012 at 6:27 PM, Rafael Espíndola <rafael.espindola at gmail.com> wrote: > I just tried asan on an optimized 32 bit build of > ------------------------------------- > #include <stdint.h> > __attribute__((noinline)) > void f(uint64_t *p) { > *p = 42; > } > int main() { > void *p; > f((uint64_t*)&p); > } >

I just tried asan on an optimized 32 bit build of ------------------------------------- #include <stdint.h> __attribute__((noinline)) void f(uint64_t *p) { *p = 42; } int main() { void *p; f((uint64_t*)&p); } ------------------------------------ and it correctly catches the invalid access. If I comment the attribute, the optimizers find and exploit the undefined behavior and

On 11/3/12 4:06 AM, Kostya Serebryany wrote: > Also note that this is not the kind of bug for which asan is good. > If we are dereferencing an uninitialized pointer, there is a high > chance that the program will SEGV w/o any tool. > If we are unlucky and the garbage is accidentally equal to some valid > address, asan will not catch it either. > Valgrind (and work-in-progress

Hi, I want to insert onto database the Current DateTime Value into database field "Store_date" with datatype (DATETIME). How to do tihs? I am usiing hidden variable into my form as follows:- ======================= <%= hidden_field("store", "store_date", :value =>:datetime ) %> ======================= But when I m submmittng form I couldn''t find entry in store_date field. How to do this usi...

Also note that this is not the kind of bug for which asan is good. If we are dereferencing an uninitialized pointer, there is a high chance that the program will SEGV w/o any tool. If we are unlucky and the garbage is accidentally equal to some valid address, asan will not catch it either. Valgrind (and work-in-progress MemorySanitizer) will catch this. --kcc On Sat, Nov 3, 2012 at 5:38 AM, Eli

I want to add virtual fields in a model. table: field1 field2 I want to add field3 which is field1*2 + field2 I know tihs can be achieved in View, but I want it in model. Then I will pass this model to PagerRenderer to paginate. How to achieve this? thanks -- Posted via http://www.ruby-forum.com/.

I recently had to uninstall PokerStars and WINE will not allow me to preform the reinstallation. I have also tried installing other programs with no success. I contacted PokerStars support but they said it'll be at least 2 days before their Linux guy was in. What they wanted was my WINE log files by going in through the terminal. Thats all they could tell me until 2 days from now. I have no

...ploaded/dir/;done?on all 14 servers. 2. uploaded file1.txt and made sure it lands on webserver1 and the same file1.txt with a minor change that landed on webserver 3. Result: 14 webservers?received?only the oldest file instead of propagating the new one. Can anyone put me on the right track with tihs? Best, --Roland?

Does anyone know how to get any of these VOIP phones to allow me to do menu selections through asterisk, like when accessing voicemail and such. Thanks :P --

Hello, I build dovecot tih --with-rawlog and created a folder dovecot.rawlog inside the root-dir of the mailbox (same level as new, cur and tmp). For testing purposes I set the env MAIL to the maildir and started libexec/imap. This generally works, but I can't find the promised logs of input and output. Where do I ha...

...se the "bindaddr" directive for both SIP and IAX2, and while SIP respects it, IAX2 doesn't. It listens on every interface, and uses every one of them for outgoing source addresses. This breaks IAX2 in my environment. Have I missed a necessary config file change, or is this a bug? -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145

Hi All, I have two Asterisks, one on the LAN that handles the internal calls with a PSTN interface and one on the DMZ with a public interface. I would like to route SIP calls from the internal users to the Internet over IAX2 to the DMZ and onwards. All users have soft phones so they would enter sip:someuser@somevoip.org to get a connection. I would like to avoid having number prefixes to dial

Hello all, I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that chan skinny is possibly capable of doing that and would like to make sure. I have P00405000600 firmware which I have put in version in skinny.conf. the phone basiclaly stops at verifying load. tcpdump shows nothing happening apart from small amount of traffic to port 2000 (skinny). Does anyone

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Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

Hello, I'm trying to migrate my system to FreeBSD and the Makefile for asterisk-addons fails in the first make clean: bash-2.05b# make clean "Makefile", line 56: Missing dependency operator "Makefile", line 57: Could not find .depend "Makefile", line 58: Need an operator make: fatal errors encountered -- cannot continue I would like to think there is no

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

what is the best way to have 2 asterisk servers communicate with each other?

Hi everybody. I'm trying to find a way to connect two (or more) extensions directly without being kept in the middle during the conversation but it won't happen. The purpose here is to have asterisk running on a low bandwidth (128Kbps) internet connection just as some kind of a proxy between some ip phones with high speed (10Mbps) internet connections. SER is not an option, for now.