similar to: [LLVMdev] llvm-build: error: invalid native target: XYZ (not in project)

You need to add your target to autoconf/configure.ac. Here are the directions from http://llvm.org/docs/WritingAnLLVMBackend.html To get LLVM to actually build and link your target, you need to add it to the TARGETS_TO_BUILD variable. To do this, you modify the configure script to know about your target when parsing the --enable-targets option. Search the configure script for TARGETS_TO_BUILD,

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

I have seen that most of the targets do comparison and branching in two separate instructions e.g. 'cmpl' followed by 'br' in x86 or the like. LLVM IR is also in same manner. I want to implement comparison+branching in one instruction like beq r1, r2, .label #if r1==r2 then jump to .label How to merge two instruction into one. Regards Vikram Singh -- View this

On Thu, Feb 13, 2014 at 5:52 PM, Renato Golin <renato.golin at linaro.org> wrote: > On 13 February 2014 13:47, Evgeniy Stepanov <eugenis at google.com> wrote: >> Hm, I see that -funwind-tables on arm-linux-androideabi target >> replaces this "cantunwind" with a proper unwind table. >> Hence http://llvm-reviews.chandlerc.com/D2762. > > If Android is

Due to an issue I'm having with 7.x, and trying to track it down, I spent tonight getting my server setup to allow my to break into the debugger when it hangs, and hopefully dump core ... But, although I *think* I've got it all, I'm obviously missing something, as it isn't breaking ... First ... I'm running a proliant server, and when I connect via SSH to ILO on that

Hi All, I seem to be having a problem with all my VSPs. When I am registering with them I don't seem to be passing my port number. This problem causes other users the inability to call my VoIP number with the VSP. My VSP showed me what they are seeing. I have changed my useragent to be: Linksys/SPA941-4.1.15 Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise.

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me.

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Hi, I am trying to do my first puppet configuration, already installed the puppetserver and client, in this link show my configuration and my puppet structure: http://paste.pocoo.org/show/212227/ But when i run the client side daemon i get this message: info: /Class[main]/Node[basenode]/Class[inittab]/File[inittab]/source: No specified sources exist err:

Dear All, I like to simulate a physiologically based pharmacokinetics model using R but am having a problem with the daspk routine. The same problem has been implemented in Berkeley madonna and Winbugs so that I know that it is working. However, with daspk it is not, and the numbers are everywhere! Please see the following and let me know if I am missing something... Thanks a lot in advance,

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP

I tried building LLVM 2.1 with no real target CPU backends enabled, just the C backend, by hacking the configure script slightly: --- /home/foad/llvm/llvm-2.1/configure 2007-09-17 22:37:52.000000000 +0100 +++ configure 2007-12-13 10:29:41.000000000 +0000 @@ -4762,7 +4762,7 @@ done ;; esac -TARGETS_TO_BUILD="CBackend MSIL $TARGETS_TO_BUILD"

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the

Is there a recommended way to install R with homebrew? Will I completely lose the GUI? .r file command editor? thanks, Ty ----- Tyler Frazier Department of Transportation Planning and Telematics Technical University Berlin http://www.vsp.tu-berlin.de/ [[alternative HTML version deleted]]

> On 18 Nov 2019, at 22:11, Peter Smith <peter.smith at linaro.org> wrote: > > On Mon, 18 Nov 2019 at 17:06, Sergej Jaskiewicz <jaskiewiczs at icloud.com <mailto:jaskiewiczs at icloud.com>> wrote: >> >> >> >> On 18 Nov 2019, at 19:55, Peter Smith <peter.smith at linaro.org> wrote: >> >> On Mon, 18 Nov 2019 at 15:23, Sergej

Hi i have the running i both sides, client and server sides the puppet 0.25.4 Get this error on server side: puppetmasterd[5363]: Report puppet_dashboard failed: wrong Content-Length format And receive this error on my client side: warning: Value of ''preferred_serialization_format'' (pson) is invalid for report, user default (b64_zlib_yaml) I am getting any reports on my

Hi, Have you try to directly describe such patterns in tblgen file? Like this: (brcond (i32 (cond_op RC:$rs, RC:$rt)), bb:$offset) MIPS backend does that. I also do this in my own backend, and seem to be working fine. On Mon, Jan 7, 2013 at 11:55 AM, Vikram Singh <vsp1729 at gmail.com> wrote: > I have seen that most of the targets do comparison and branching > in two separate

Hi I am trying to setup asterisk at home, I have 1 in bound VSP (I have a register cmd setup for that in asterisk). At home I have a cordless phone with 2 line capability - I currently have 2 spa3102's in place to handle the 2 lines ( I am in the process of buying tdm410 to handle to handle this and the backup pstn line). I also have 2 laptops setup with soft sip phones. What I would like