similar to: [LLVMdev] llvm-ld with -O params?

I'm also running into an issue with llc. Alchemy is trying to use -march avm2 and other avm2 options, but: llc: for the -march option: : Cannot find option named 'avm2'! llc: Unknown command line argument '-avm2-use-memuser'. Try: 'llc --help' llc: Unknown command line argument '-avm2-package-name=cmodule'. Try: 'llc --help' Was avm2 support dropped?

On 2009-07-22 22:14, Simmons, Aaron wrote: > > I’m also running into an issue with llc. Alchemy is trying to use > –march avm2 and other avm2 options, but: > > llc: for the -march option: : Cannot find option named 'avm2'! > > llc: Unknown command line argument '-avm2-use-memuser'. Try: 'llc > --help' > > llc: Unknown command line argument

My apologies. I didn't realize that Adobe had branched llc. What about llvm-ld and -O? -----Original Message----- From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] On Behalf Of Török Edwin Sent: Wednesday, July 22, 2009 13:41 To: LLVM Developers Mailing List Subject: Re: [LLVMdev] llvm-ld with -O params? On 2009-07-22 22:14, Simmons, Aaron wrote: > > I’m

Eeeeck! Asterisk is inserting random digits in dialed numbers. So far I've seen it insert a 2 after the STD (area) code and insert an extra 6 or 7 in the STD code. It's pretty repeatable although the inserted number changes. My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). There's an ISDN PBX on the second span and a BRI euroisdn on the first. Calls from the

Oct 21 10:04:05 alchemy dovecot: imap-login: Login: user=<paulproteus>, method=PLAIN, rip=127.0.0.1, lip=127.0.0.1, TLS Oct 21 10:05:44 alchemy dovecot: IMAP(paulproteus): file message-decoder.c: line 289 (message_decode_body): assertion failed: (pos < input->size) Oct 21 10:05:44 alchemy dovecot: IMAP(paulproteus): Raw backtrace: imap [0x80c8e10] -> imap [0x80c8d1c] ->

I 'hg pull'ed and now my Dovecot segfaults when I select a particular folder. I've removed all files with 'index' in the name, so I don't think it's due to bad indexes. Oct 21 09:18:40 alchemy dovecot: Dovecot v1.1.beta3 starting up Oct 21 09:18:47 alchemy dovecot: imap-login: Login: user=<paulproteus>, method=PLAIN, rip=127.0.0.1, lip=127.0.0.1, TLS

Jerry, Jeremy, et. al. Last night's upgrade from cifs-mount-3.2.0-24.1.121 to cifs-mount-3.2.0-24.1.122 on a openSuSE 10.3 x86_64 machine broke mount -t cifs. Now attempting to mount shares results in the following: 11:10 alchemy/home/david/bin] # mount.cifs -v //nirvana/david /mnt/nirvana-david -o username=david,uid=1000,password=notmypass parsing options:

Has anyone managed to get Asterisk to work with any of the Alchemy or Cybergear ISDN switches? Cheers Matthew

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) -

All, If the post "Samba 3.6 -> 4.1 change that prevent sharing / - mount error(13)" makes it to the list, I apologize, but it has been over a day and I have no indication that the post ever made it to the list. The issue is that in versions prior to 4.x, I was able to create a root share in samba, but now, something prevents it from being mounted. The relevant smb.conf is:

I use realtime for my Asterisk configuration and are now making the transition to Asterisk 13 and PJSIP. I used alchemy to set up my databases and I can now configure my endpoints. While trying to configure a trunk I can see that there is a database table called ps_registrations that should be used to make the registration to the provider but there is no corresponding entry in the

Tried to submit this email this morning and didn't see it in the list. I apologize if it is a dupe. I've inherited a customized Asterisk installation. After the past time change all clocks in my office are behind by one hour. After some digging it appears we have: A /tftproot/sip.conf that is being pushed out to our phones. I found the following line that seems to be what controls

Once in a while, someone mis-dials when transfering a call on their Snom SIP phone (using the Transfer button). Instead of sending them to, say, 1940; they mistype and enter 194 or 190 or somesuch. This ends up on the PSTN (for which three digit calls are valid); not what anyone wanted. On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went

Hey, all. I'm rather new to wine development, so please point me at the right place to post diffs if this isn't it. This patch implements RtlGetNtVersionNumbers, called by lots of windows XP apps (including winver, cmd.exe, and regedit). BTW, are we just not bothering to match ordnals on our exports, or is there some deeper reason for using @s? As the patch says, the API is

Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

http://bugs.freedesktop.org/show_bug.cgi?id=16533 Summary: Vimeo does not work :) Product: swfdec Version: 0.7.x Platform: x86 (IA32) URL: http://www.vimeo.com/1026973 OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: plugin AssignedTo: swfdec at

http://bugs.freedesktop.org/show_bug.cgi?id=15623 Summary: Support for Vimeo.com Product: swfdec Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: enhancement Priority: high Component: plugin AssignedTo: swfdec at lists.freedesktop.org ReportedBy:

I have a legacy ISDN PBX (Network Alchemy Argent Office) connected to Span 2 of a Digium Wildcard TE205P. Recently Calls from this PBX have been failing with ISDN Cause Code 38 (Network Out of Order!). The problem seems to be getting worse and is now effecting more calls than not (although this could just be because I'm aware of it). Once the ISDN PBX has decided the Network's Out of