search for: codecompletion

greened at obbligato.org (David A. Greene) writes: > For now, I think if I tweak the way I do the build to always build > without pointing to llvm-gcc first, build and test LLVM then build > llvm-gcc and re-build LLVM, it should work. It will take much longer, > though. :( I updated the bug explaining what I'm seeing. I think the correct fix is to use absolute paths to tools

...ub -all {([^a-zA-Z_-])opt } $new_line "$llvmtoolsdir/\\1opt " new_line regsub -all {opt } $new_line "$llvmtoolsdir/opt " new_line regsub -all {tblgen } $new_line "$llvmtoolsdir/tblgen " new_line It appears to work for clang tests: [x86_64-off-opt]: FAIL: Clang :: CodeCompletion/ordinary-name.c (432 of 8411) [x86_64-off-opt]: ******************** TEST 'Clang :: CodeCompletion/ordinary-name.c' FAILED ******************** [x86_64-off-opt]: Script: [x86_64-off-opt]: -- [x86_64-off-opt]: /ptmp/dag/build.llvm.trunk.official.opt/x86_64-unknown-linux-gnu/Release+Asserts/b...

Jason Kim <jasonwkim at google.com> writes: >>> There is definitely something to this.  If I take a random failing >>> testcase and run the test in isolation in the shell, it works.  So >>> what, if anything, does lit/FileCheck/etc. do that might run >>> interference if there is another copy of lit/FileCheck/etc. running >>> at the same time? I

Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a

Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this, I'd like to know which directories/files are required for a basic install? Does this look right? ================= /bin/asterisk /etc/asterisk/ asterisk.conf

Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you.

Hello I need to find a recent and neutral comparison of the major products available to connect an Asterisk server to the telephone network, whether ISDN (BRI) or PSTN, and through a PCI card or some external box. I'm told there are less issues (echo, stability) with external boxes compared to PCI cards. Apparently, the main brands are Digium, Sangoma, Rhino Equipment, Patton, and

Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you.

...kup/ParameterizedMethodBinding.<init> CompilerOracle: exclude org/eclipse/cdt/internal/core/dom/parser/cpp/semantics/CPPTemplates.instantiateTemplate CompilerOracle: exclude org/eclipse/cdt/internal/core/pdom/dom/cpp/PDOMCPPLinkage.addBinding CompilerOracle: exclude org/python/pydev/editor/codecompletion/revisited/PythonPathHelper.isValidSourceFile CompilerOracle: exclude org/python/pydev/ui/filetypes/FileTypesPreferencesPage.getDottedValidSourceFiles Has anyone else seen this? Any idea what's wrong? Is anyone here are using Eclipse for C++ development, anyway? Eclipse platform version is 3...

Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hello I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to call 3. Asterisk puts me on hold through Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the

Hello About every three months, my dad's little Asterisk server that handles his business phone line with an OpenVox PCI card stops taking calls. To check if it's the cause, I'd like to run a CRON job every night to restart Zaptel and Asterisk. Before I go ahead, I'd like to know if I can just send the following commands, or if there are issues I should know about:

Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

PaulR (sorry again if this is known knowledge) > There's no reason for Herald to be adding project LLVM/subscriber llvm-commits at the last second here. Its possible the rL (LLVM) had be added as the repository in the review on creation rather than rCFE, if thats the case then the herald rule "H270" is going to fire because it see the repository in the review, so add LLVM

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hello I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. 1. After successfully running "make all; make install; make config", I edited /etc/dahdi/system.conf thusly: loadzone=fr defaultzone=fr fxsks=1 2. Then ran "dahdi_cfg -vv" which says: ------------- DAHDI Tools

Hello In case Asterisk is used in a private LAN behind a firewall while allowing remote SIP clients to connect from the Net, we must open UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let incoming voice packets. Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP...