similar to: [LLVMdev] Bug in DragonEgg or LLVM

Hi, I was wondering whether or not MPI-libraries are expected to work with llvm-gcc? I tried to compile openmpi-1.3 using the llvm-gcc4.2-2.5-x86-darwin9 distribution on my MacBook Pro running OS 10.5.6. Installation using the gcc (gcc version 4.0.1 (Apple Inc. build 5490)) worked well. But for the llvm-gcc I get the following error message (during the make): > .... > Making all

Yes I think I can. The original compile line was: llvm-g++ -DHAVE_CONFIG_H -I. -I../.. -I../../extlib/otf/otflib -I../../ extlib/otf/otflib -I../../vtlib/ -I../../vtlib -D_REENTRANT -fopenmp - DVT_OMP -O2 -MT vtfilter-vt_tracefilter.o -MD -MP -MF .deps/vtfilter- vt_tracefilter.Tpo -c -o vtfilter-vt_tracefilter.o `test -f 'vt_tracefilter.cc' || echo './'`vt_tracefilter.cc I

Wojciech, I've just commited a patch to llvm-gcc 4.2, which moves openmp lowering stuff to be run little bit earlier, so llvm-convert will catch its result. It looks now gcc atomic & sync builtins should be introduced to llvm as a remaining ingredient. Example program from Diego's paper now compiles to: @.str = internal constant [10 x i8] c"sum = %d\0A\00" ;

We are running 1.2.14 with an IVR in the dialplan. If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a couple of digits very rapidly (I found this with 33 on a sticky keypad) which are an invalid response, Allison will go into a loop saying 'I'm sorry, that is an invalid response, please try again.' over and over. This does not happen with a commercial

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Hello, I want to find out the member variable of the structure (.omp_data_i) being accessed but the metadata and the use don't seem to be linked. Here's is a portion of the IR generated for a function containing an OpenMP pragma: define internal void @bpnn_adjust_weights.omp_fn.0(i8* nocapture %.omp_data_i) nounwind uwtable { entry: %0 = getelementptr inbounds i8* %.omp_data_i, i64

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in

On 04 Aug 2014, at 09:27, Reid Kleckner wrote: > On Sat, Aug 2, 2014 at 7:51 AM, Jonas Maebe > <jonas.maebe at elis.ugent.be> > wrote: > >> On 01/08/14 19:37, Reid Kleckner wrote: >> >>> What happens if you drop appending linkage? I think it will just >>> work, >>> since you are already using a custom section, which will ensure

Having some issues with getting sound from a call. I have 4 systems. 3 main systems which handle calls for our 3 locations. The 4th system is the central voice mail system. When an inbound call gets passed to someones voice mail its done with an IAX2 connection. The same happens after hours when we have our night mode set. If you dial the main number after hours you are passed straight to the

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]

On 11/02/2011 11:17 AM, Marcello Maggioni wrote: > Mmm I found out a very strange behavior (to me) of the SCEV analysis > of the loop bound of the external loop I posted. > When in ScopDetection it gets the SCEV of the external loop bound in > the "isValidLoop()" function with: > const SCEV *LoopCount = SE->getBackedgeTakenCount(L); > > It returns a

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]

OK, this is how I thought it's supposed to work. It just confounded me why the book would say the Playback() and Background() syntax were the same, then in the very next paragraph give an example that belied that claim. On 6/17/2023 1:46 PM, Doug Lytle wrote: > On 6/17/23 08:47, Steve Matzura wrote: >> >> Both Background()  and WaitExten()  allow the caller to enter DTMF

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : ==== Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command' Apr 30

Hi, > The following code using OpenMP pragmas , when compiled with gcc 4.5 + LLVM 2.8 + DragonEgg 2.8 and ran, produces segmentation fault. I just tried it on x86-64 linux with latest dragonegg from svn, and it executes fine, with no valgrind complaints. After compiling at -O0 valgrind does say ==7251== Syscall param exit_group(status) contains uninitialised byte(s) ==7251== at

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hi all, I have just setup a trixbox system and I am implementing hylafax/iaxmodem solution for the faxing. When i send a fax to it by phoning in listening to the IVR and manually pressing start to initate the fax, the call gets picked up correctly as a fax and everything works well. When I try sending a fax by entering the phone number and pressing start to initiate dialing it sounds like