search for: alchemi

I am using LLVM via the Adobe Alchemy project. I'm trying to upgrade LLVM within Alchemy (its v2.1 of opt is crashing), but am running into an issue with llvm-ld. Alchemy is passing -O5 to llvm-ld, but it seems that -O params are no longer accepted by llvm-ld v2.4+ ("llvm-ld: Unknown command line argument '-O5'. Try: 'llvm-ld --help'"). Are -O optimizations

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.

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?

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

Adobe has created an LLVM based C to Flash VM compiler. Their example code includes the Vorbis library: http://labs.adobe.com/wiki/index.php/Alchemy:Libraries This should make producing a Flash Ogg/Theora player using the Theora reference library a fairly straight forward process. Such a player will never perform as well as native support, but it would be highly valuable along side cortado

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] ->

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

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

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

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:

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:

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

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

Listmates I'm running rsync 3.02 and I have a simple thing I'm trying to do, but rsync won't let me do it. What I'm trying to do is consolidate saved rpms on openSuSE 11.0 to a central server for local updates. openSuSE saves the rpms in: /var/cache/zypp/packages/<repository name> I have written a script that parses all the file names and creates temp files that holds

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

Hi, I am using samba 2.2.7, configured with winbind, connected to a domain. I have publically shared printers that are done via user map. I also have many shares that I need to authenticate for, due to access rights. With one of the printers installed on my XP laptop, if I try to connect to a share, but inputting my "DOMAIN\user" and "password" I get the following error:

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