search for: stever

Hi, We needed a vorbis decoder that operated differently from the approach taken libvorbisfile. So I sat down, looked through the docs, and the sample decoder that come with libvorbis and wrote our own. Since I couldn't find any docs for the API for libvorbis itself, I've written some. It details all the steps you need to go through to write a _basic_ reusable decoder library using

I am giving serious consideration to abandoning waveOut multimedia and the wav format, for most of the audio in my C++/MFC project. However, I've never done any programming with a codec, and I still can't see how to step into Ogg Vorbis. If I convert all my wav files to ogg, there seem to be two major issues I'm facing: 1) What exactly will I replace my waveOut functions with? 2)

I am giving serious consideration to abandoning waveOut multimedia and the wav format, for most of the audio in my C++/MFC project. However, I've never done any programming with a codec, and I still can't see how to step into Ogg Vorbis. If I convert all my wav files to ogg, there seem to be two major issues I'm facing: 1) What exactly will I replace my waveOut functions with? 2)

Greetings - I have not been a long time follower of this list, but I have scanned through the last year or so of archives, after not finding much from google searches. I am hoping someone here can inform me if what I want to do is feasible, and give me some general guidance to follow so that I can continue my research and complete this task. I admin a RH3 system that is primarily a Samba

Forwarding a message from Steve \"Stevers!\" Coile: > This is already possible. Create the file then unlink it. The file > contents aren''t deleted and i-node isn''t released until the file is > closed. Without kernel support, unlinking the file at creation time requires a code change in the process which...

I am running function tests to see if ActionMailer mails out an email. I have two development machines, both with rails 1.1.6. On one machine, the ActionMailer::Base.deliveries array resets itself after each test. On the other machine, the array keeps accumulating the emails from the previous function tests and doesn''t clear out the array after each test. Both machines have identical

So I want to use facts from one host in the configuration of other hosts. Trivial example: setting restrict lines in ntp.conf to allow a monitoring host to query ntpd on remote hosts. So if the monitoring host boots with IP 10.2.2.3 (which may change regularly), and it knows its the monitor host (via some customer facter plugins), I want some global variable to be available

>Forwarding a message from Steve \"Stevers!\" Coile: >> This is already possible. Create the file then unlink it. The file >> contents aren''t deleted and i-node isn''t released until the file is >> closed. > >Without kernel support, unlinking the file at creation time requires >a code chan...

/ I'm testing a new setup of postfix using deliver to put the messages into the user's maildir, and am curious if there is a way to make deliver create custom maildirs when it delivers to a valid user without a maildir? Right now it only creates the inbox, I'd like it to create the .sent/ .Trash/ and .SPAM dirs. Is this possible? Thanks -------------------------------------- /

As most others have - I''ve been having issues with dedup. Here''s my situation, 4TB pool for daily backups of sql server - dedup enabled - so a typical directory has 100+ files that are mostly identical (some all are identical). If I do rm * OpenSolaris is dead, zfs hung, etc. sometimes it comes back after hours, sometimes I reboot (only if a directory I don''t care if

All, I was curious as to what size of systems (cpu, memory) that you are using. I'm going to build a new server for imap, imap-ssl, pop3, pop3-ssl, maildir storage (NOT smtp or spam --that's on another server) for ~500 users. What do you believe is the minimum for this? Thanks, Steve

We want to test capacity of our MeetMe room. The thing that is distinct about this is that the incoming line is being delivered IAX2 to our server across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test both the VoIP Gateway and the MeetMe room performance. You can reach our MeetMe room directly at 1-301-561-9229 If you want to test with us we're thinking maybe 9pm

I'm guessing you aren't on a digital line (i.e. POTS). If that's the case you'll have problems sometimes that the dial tone isn't there as fast as it dials. I forget the pause character (p? It seems like that was pulse not pause). I'd break this into 2 steps, answer and announce something - make sure you are answering fine. Step 2 - Dial via ZAP and hopefully then you

I'm having a problem with Caller ID matching. The call is coming in via IAX2 to our system, the caller id doesn't seem to parse right. I just got the latest CVS version an hour ago or so. Relative extensions are pretty simple: [disaid] ; ; Check caller id for disa access ; exten => s,1,Wait,0 exten => s/7031234567,1,goto,disa|s|1 exten => s,2,congestion [main] exten =>

Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible).

[Mod: redirected to linux-security --alex] On Mon, 24 Mar 1997, Alexander O. Yuriev wrote: >This is yet-another reason to _partition_ your disks. Of course hard links >do not work accross filesystems. Even thought it is a pain in the neck to do >when installing your operating system, think about separating critical >system files from non-critical and non-system files from system

I notice most things say to open ports 10000-20000 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) This call has no audio, presumably because port 8542 is firewalled in the iptables on the server.

I'd like to integrate a few peoples cell phones into our asterisk system as we're walking around a data center some days and carrying another cordless phone just doesn't seem to make sense. Forwarding is easy, however once forwarded I want to be able to flash and transfer back to another party / voice mail / etc. How can this be done? As far as I know there's no way to generate

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

...test "obvious" build I see. Am I just not RTFM enough on finding new builds? I hate to move to Nexenta, I would think in the future Oracle will maintain this better than a third party and don''t want to switch back and forth. Steve Radich - www.BitShop.com - www.LinkedIn.com/in/SteveRadich -- This message posted from opensolaris.org