similar to: [LLVMdev] Metadata nodes and C API

On 4 September 2012 13:44, Anthony Bryant <antjbryant at gmail.com> wrote: > Hi, > > I'm relatively new to using LLVM, and I'm working on a front end for a > language I'm designing. I have a couple of questions about metadata: > > 1. I'd like to be able to store some high level program structure in > bitcode files, to be imported by future runs of my

Hi Anthony, You need to send patches to the llvm-commits mailing list to get them reviewed and committed. -Dave -- Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation > -----Original Message----- > From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] > On Behalf Of Anthony Bryant > Sent: Wednesday, September

Greetings, I'm trying to implement exception handling in my front end. I have a prototype working using the Itanium ABI on Linux x86_64, but I'm not sure how to proceed for other platforms. So I was wondering: which OS/architecture combinations support zero-cost EH, and which support SJLJ? My main concern is keeping the IR for call sites platform independent. Right now, it looks like to

Hi Bill, Thanks for the response, I just have a couple more questions. On 11 February 2013 06:45, Bill Wendling <wendling at apple.com> wrote: > As far as which OS/architectures support zero-cost EH and which > support SJLJ, LLVM is designed to assume that Intel supports zero-cost > exceptions and ARM supports SjLj exceptions. However, it's all in the > personality function

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

On Feb 9, 2013, at 4:14 PM, Anthony Bryant <antjbryant at gmail.com> wrote: > Greetings, > > I'm trying to implement exception handling in my front end. I have a > prototype working using the Itanium ABI on Linux x86_64, but I'm not > sure how to proceed for other platforms. > > So I was wondering: which OS/architecture combinations support > zero-cost EH,

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38

Hi , I am getting following error when i try to run xenoprof /usr/local/bin/opcontrol: line 1082: echo: write error: Invalid argument Using 2.6+ OProfile kernel interface. Reading module info. Using log file /var/lib/oprofile/oprofiled.log Daemon started. what is this write error ? Any help is really appreciated. -John _______________________________________________ Xen-users mailing list

Hi , I am getting following error when i try to run xenoprof /usr/local/bin/opcontrol: line 1082: echo: write error: Invalid argument Using 2.6+ OProfile kernel interface. Reading module info. Using log file /var/lib/oprofile/oprofiled.log Daemon started. what is this write error ? Any help is really appreciated. -John _______________________________________________ Xen-users mailing list

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant

My setup is Samba/smbclient version 3.0.1 on linux RedHat 9. \\snapper\dfs is a dfs share on snapper, a Win2k3 Active Directory server. My smb.conf contains: [global] realm = ciswinnet.cis.uab.edu workgroup=CISWINNET security = ADS encrypt passwords = yes password server = snapper.cis.uab.edu client use spnego = yes I am trying to run the following command: smbclient \\\\snapper\\dfs

deeper still looks like culprit was fts_enforced = "no" which i removed given default anyway .. still get error [root at stargate dovecot.rawlog]# doveadm search -u matt at the-bryants.net mailbox .EXPUNGED/* from noreply at apress.com doveadm(matt at the-bryants.net): Error: fts: Failed to sync mailbox .EXPUNGED: Mailbox doesn't exist: .EXPUNGED 6156a416748c035b5c0b0000ef794dd1

So virtual/combined was meant to combine INBOX/Sent/Drafts and virtual/all display them in a threaded manner .. at least that was the intention ... rgds Matt > Aki Tuomi <mailto:aki.tuomi at dovecot.fi> > 18 February 2018 at 11:10 pm > > Uhm. > > combined folder seems to use INBOX for saving mails and contains mails > from Sent and Drafts. > > Not sure what your

Ok I can't get my CDR values to set from the h extension in either 1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf ; Normally, CDR's are not closed out until after all extensions are finished ; executing. By enabling this option, the CDR will be ended before executing ; the "h" extension so that CDR values such as "end" and "billsec" may

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components: