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Hello, using the LLVM API, I've build one very simple function that adds two ConstantInts and returns the result. I noticed that, when I emit IR code, it is optimized to a simple "ret i16 42" when I add 40 and 2. I'd like to see the operations that are necessary to compute the result, though. Can I somehow disable this optimization in the pass, leading to more verbose IR code?

IRBuilder is a templated class, and one of the template arguments is the constant folder to use. By default it uses the ConstantFolder class which does target-independant constant folding. If you want to disable constant folding you can specify the NoFolder class instead, i.e. declare the builder as follows: IRBuilder<true, llvm::NoFolder> builder(body) On 26 Nov 2013, at 19:23, Daniel

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

I am trying to code a Web Service in rails that can handle file uploads. Now as i read, SOAP 1.1 doesn''t support this yet..so i will have to use SOAP4R, right? This is fine from server side...but will this API will be compatible to .NET clients, which will be using standard SOAP API perhaps. Any idea, whats the way to go here?

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

Thanks Jud, I was able to re-encode my entire FLAC archive into 1.1.4, although on my slow computer (with all other apps closed) it took over 3 days to complete! Only one song returned an error and that was quite easy to fix. I did run into one issue though. I have a FLAC album in which the songs are all 24/96 and REFLAC fails when it runs. I want to re-encode them from compression level 5

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

I have begun testing dovecot as (primarly) a POP replacement. So far the only lacking bit is the logging. It would be great if pop3 could log a line at client exit with the following info: (wishlist follows :) o nr deleted mails + total bytes deletes o nr mails left in spool + bytes left o time spent o indication if TLS was used or not If you would lika generic logging API I have one you can

I MAY be able to derive something out of the present example given here : http://blog.codeinmotion.com/index.php/2006/12/22/pdf-generation-in-ruby-on-rails/ but this deals with filling out forms. Is there a simpler example that just allows you to talk to itext , send it some plain text and get back a pdf and then send that pdf to the user as downloadable / renderable data? -- Posted via

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

Hi all, As mentioned earlier, we've been working on getting LLVM to build on Solaris on x86/x86-64. The changes required turned out to be fairly minor, so it's perhaps a bit of a stretch to call it a "port". There were two main issues that we ran into here: 1. The Solaris x86 ABI by default defines the x86 registers CS, DS, ES, etc in the system headers, which

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

Hi, I have a difficulty on my Asterisk's database.How can I get the info about list of ringing agents on my queue In console : -- Started music on hold, class 'default', on DAHDI/77-1 *-- SIP/6002-00cc0f90 is ringing -- SIP/6004-00c23270 is ringing -- SIP/6005-00be6220 is ringing* -- SIP/6004-00c23270 answered DAHDI/77-1 -- Stopped music on hold on DAHDI/77-1

Hi, I've got a MSAccess database on a linux server (fedora core 5, samba 3, 192.168.0.90) which a Win XP client machine needs to use (192.168.0.50). However, Access refuses to open the database because "it's not on the LAN so is a security risk". How can I configure Samba so that Windows recognises 192.168.0.90 as a local machine, please? Jim Donovan Office +61+2-8923-5208

On Thursday, December 19, 2002, at 06:10 am, Monty wrote: > Het rillian: How to get HTML output from the XML in order to roll > changes back to the website? Or do you plan to replace the HTML > on-site too? Sorry, I got distracted before I finished everything. If you have the tools installed (xsltproc, passivetex) 'make' should generate monolithic pdf and html

Hi After upgrading from 1.4 to 11.4.0, I *am* able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit <salah.elharit200 at gmail.com> wrote: > hello list, > > i have asterisk 11.15.0 and i have some trunks sip from my provider > > we have some ip phone astra 6731i > > each Ip-phone is configured with trunk and we call > > no ihave configured another trunk from the same provider in my asterisk > > i can call

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records despite the dialplan setting it. My program generates the calls by setting Channel=Local/NUMBERTODIAL at