similar to: Can YOU find a trailing parenthesis?

Hi Kirill, Don't forget to add X86TargetMachine.obj (add to Additional Dependencies in Linker options, if you are using MSVS) otherwise LLVM will try and use Interpreter instead of JIT. Hope this helps, Rob. > -----Original Message----- > From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] > On Behalf Of kirill havok > Sent: Wednesday, August 20, 2008

kirill havok wrote: > Hi Gordon, > I wrote a small example, but while running I get an error("Tied to > execute an unknown external function"), where I am wrong? > I think the problem is that some_test_func() is a C++ function, so its name is being mangled during compilation. To fix it, I think you want to add a declaration telling the compiler to treat some_test_func()

Hi Gordon, I wrote a small example, but while running I get an error("Tied to execute an unknown external function"), where I am wrong? Thanks in advance. Kirill. int some_test_func( int ){ std::cout << "!!!!" << std::endl; return 8848; } int _tmain(int argc, _TCHAR* argv[]){ Module *M = new Module("test"); ExistingModuleProvider* MP = new

On Aug 19, 2008, at 13:36, kirill havok wrote: > I got very interested in LLVM project, and I decided to start > writing my own scripting language based on it. By studying the > documentation, I could not find how to call external function, > written in C. That is, I have a set of functions written in C/C++, I > generate code, using LLVM C++ interface, how can I call(or

Hello, I got very interested in LLVM project, and I decided to start writing my own scripting language based on it. By studying the documentation, I could not find how to call external function, written in C. That is, I have a set of functions written in C/C++, I generate code, using LLVM C++ interface, how can I call(or register in machine in run-time) my external functions? Maybe I just missed

JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Hello, I'm just wondering if anyone has fixed the 'InUse&Ringing' problem. * v1.4.23.1 Ta

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted If (${ael-var} = 1) { Playback(beep);

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

Hello all, I have being trying to replicate the following call scenario with my Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html <http://www.tech-invite.com/Ti-sip-service-8.html> I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open

https://bugzilla.samba.org/show_bug.cgi?id=13433 Bug ID: 13433 Summary: out_of_memory in receive_sums on large files Product: rsync Version: 3.1.3 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: core Assignee: wayned at samba.org Reporter:

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

I would like to know if I can set Call Forwarding on an extension but allow direct calls from the extension it is forwarded to. Example: Extension 100 sets call forwarding (all) to extension 101. All calls to 100 are immediately forwarded to 101 as expected. However, if 101 tries to transfer a call to 100 or tries to call 100 directly, it sounds "busy" because it obviously goes into