similar to: MD5 in SIP's "register => ..."

Hi all, when compiling code like short ptr * = some_address; int val; val = *ptr; if (val>2047) val = 2047; else if (val<-2048) val = -2048. // other things done that require val to be an int ... The load operation is represented by a load and a sign extension operation in the LLVM IR. On most target architectures, there exist signed halfword load instructions, so the load and

Hello folks, I was trying to set up Dovecot deliver LDA instead of Postfix default virtual LDA In Postfix main.cf I specified: virtual_transport = dovecot dovecot_destination_recipient_limit = 1 In Postfix master.cf I entered: dovecot unix - n n - - pipe user=mailer flags=DRhu argv=/usr/local/mail/libexec/dovecot/deliver -f ${sender} -d ${recipient} In

On 21 Jan 2013, at 14:39, Justin Holewinski <justin.holewinski at gmail.com> wrote: > Instruction selection happens on a different IR: SelectionDAG. In this IR, there are sign-extending loads that the IR converter will use, and are used for example to load 8/16-bit values into 32-bit registers (with sign or zero extension). Any optimizations performed during codegen will be in this

http://bugzilla.mindrot.org/show_bug.cgi?id=948 ------- Additional Comments From dtucker at zip.com.au 2005-01-19 20:01 ------- Also worth trying: patch #772 in bug #973 ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.

Instruction selection happens on a different IR: SelectionDAG. In this IR, there are sign-extending loads that the IR converter will use, and are used for example to load 8/16-bit values into 32-bit registers (with sign or zero extension). Any optimizations performed during codegen will be in this representation, or even MachineInstr form, which is post-isel and any sign-extension information

On Jan 21, 2013, at 6:34 AM, Justin Holewinski <justin.holewinski at gmail.com> wrote: > > On Mon, Jan 21, 2013 at 9:16 AM, Bjorn De Sutter <bjorn.desutter at elis.ugent.be> wrote: > On 21 Jan 2013, at 14:39, Justin Holewinski <justin.holewinski at gmail.com> wrote: > >> Instruction selection happens on a different IR: SelectionDAG. In this IR, there are

On Mon, Jan 21, 2013 at 9:16 AM, Bjorn De Sutter < bjorn.desutter at elis.ugent.be> wrote: > On 21 Jan 2013, at 14:39, Justin Holewinski <justin.holewinski at gmail.com> > wrote: > > Instruction selection happens on a different IR: SelectionDAG. In this > IR, there are sign-extending loads that the IR converter will use, and are > used for example to load 8/16-bit

OK, here is the deal. My system spins like a top with one exception. I would like to store Computer accounts in ou=Computers and get them out of ou=People. I've tried simply makeing this the ou that Computers are added to but then the XP clients cannot seem to see them. Changeing this is likley to be easy to do and incredibly hard to find the specific details of how to do it. I know

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

Just a little pun there! I've been mostly lurking for a couple of weeks and realize how little I know and understand about this PBX and phone stuff. I did a little looking about and came across a glossary but they terms are -- for me -- kind of out of context. I'm wondering if there is (much as I hate the term "Dummy") a "PBX for Dummy's" or similar. I've

Hello! I have a few sip devices and it is necessary for me to disable call-waiting and immediately return a busy signal if the sip's channel is busy on them. What is the procedure to do so in Asterisk 1.4? Thank you, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to find a way to call voicemail without being prompted for my mailbox number. I was wondering if there was a variable for sip mailbox, or is there a way to define a variable that matches a sip's mailbox. I tried using "exten => 996,1,voicemailMain(${CALLERIDNUM})" but this only works if the mailbox matches the caller id. Any suggestions would be appreciated.

Hello, all. I am working on a project with a large (~350Mb, about 5800 rows) insurance claims dataset. It was supplied in a tilde(~)-delimited format. I imported it into a data frame in R by setting memory.limit to maximum (4Gb) for my computer and using read.table. The resulting data frame had 10 bad rows. The errors appear due to read.table missing delimiter characters, with multiple data

Hello, all. I ran across an odd problem while working in R 2.6.0. The command line text follows. Basically, I attempted to convert a character vector of length 13 (in a data frame with 13 rows) from a character representation of dates to a POSIX representation using strptime. strptime returned a vector of length 9, which appears to contain 13 values (!) in the appropriate format. I can't

Well I'm slowly learning my way around asterisk although as yet I haven't had the chance to actually hook the system up to an ISDN line. I am going to migrate from an Argent Office setup. My only problem is keeping costs down on the phones. The Argent system is running about 30 POTS phones. Can someone suggest the cheapest option? Should I get some kind of large scale FXS box or would

I'm now trying to use an m3u formatted list to pass to ezstream rather than stdin. Since all of my audio files are in .wav format, and I have oggenc, I see no reason to decode them before ogg encoding them. Is there a configuration I can use to make this happen? I tried omitting the decode parameter and ezstream complained. Here is what I have now: <ezstream>

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

I'm now trying to use an m3u formatted list to pass to ezstream rather than stdin. Since all of my audio files are in .wav format, and I have oggenc, I see no reason to decode them before ogg encoding them. Is there a configuration I can use to make this happen? I tried omitting the decode parameter and ezstream complained. Here is what I have now: <ezstream>

Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone