similar to: Asterisk CVS stable (current) crashes on remote user (over CAPI) pressing # or * when in conference

hi! i send again my question because there was a problem earlier that someone did not see my attached file. If you really can't download it, this is the attached file. Please help me how to convert this S plus file to R. Is there a quick method to do it? I don't have an S plus installer here. --------------------------------------------------- # Computes a possible choice for

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hey, I have a data frame x which consists of say 10 vectors. I essentially want to find out the best fit exponential smoothing for each of the vectors. The problem while I'm getting results when i say > lapply(x,ets) I am getting an error when I say >> myprint function(x) { for(i in 1:length(x)) { ets(x[i],model="AZZ",opt.crit=c("amse")) } } The error message is

Asterisk Project Security Advisory - ASA-2007-014 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Stack buffer overflow in IAX2 channel driver |

Asterisk Project Security Advisory - ASA-2007-014 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Stack buffer overflow in IAX2 channel driver |

Hi, A potential client (hotel) has a Property Management System that talks the "Mitel" protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or better still have experience interfacing asterisk in a hotel situation like this? The PMS

hello to all, i am getting weird syntax error while running the rake db:migrate syntax error, unexpected tSYMBEG, expecting kDO or ''{'' or ''('' Apparently i am creating an engine in my main application using ''ruby script/generate plugin'' command in rails 2.3.0 with engine having its own separate database define in database.yml of main

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Thank you very much Dirk! Le mer. 13 mai 2020 ? 14:59, Dirk Eddelbuettel <edd at debian.org> a ?crit : > > Salut Adrien, > > It appears to be a bad OpenMP and and OpenBLAS interaction you can (for > now) > avoid) by replacing the 'pthread' variant of OpenBLAS with the OpenMP > version > (see the thread for details). Doing > > sudo apt install

Howdy, Has anyone worked on a Mitel-2000 emulation for PMS integration (Hotel mgmt systems)?  Hoping to get my hands on the protocol definition (RS-232!!) for check-in/check-out/housekeeping/CDR, but if someone has already done I would totally buy it. Cheers, -- Jeff LaCoursiere StratusTalk, Inc.

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

"A module pass can use function level passes (e.g. dominators) using getAnalysis interfacegetAnalysis<DominatorTree>(Function), if the function pass does not require any module passes." http://llvm.org/docs/WritingAnLLVMPass.html In your case, A module pass (ModPass2) is trying tu use function level pass (FunPass1) which uses module level pass (ModPass1). This is not

Having a ModulePass that requires a FunctionPass that in turn requires another ModulePass results in an assertion being fired. Is this expected behavior (that seems to be undocumented), or a bug? Specifically, the following code will emit the assertion: [VMCore/PassManager.cpp:1597: virtual void llvm::ModulePass::assignPassManager(llvm::PMStack&, llvm::PassManagerType): Assertion

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

Both updated from CVS within the same script but:- channel.c:44:2: #error "You need newer zaptel! Please cvs update zaptel" channel.c: In function `ast_channel_alloc': channel.c:298: error: `ZT_TIMERPONG' undeclared (first use in this function) channel.c:298: error: (Each undeclared identifier is reported only once channel.c:298: error: for each function it appears in.)