similar to: flac for Mac

Hi - I am currently developing a RTSP/RTP/SDP solution to stream Speex encoded data. Using my current source, I have successfully streamed u-law and PCM encoded audio but have been unsuccessful thus far with Speex. Because of some constraints of my system, I am encoding audio at 11.025kHz. I am still using the 160 samples per frame which makes my frame size 28 bytes. I have successfully

> I believe there is new resampling functionality in the speex svn head, > although I haven't tested it yet. You might also want to check out > 'Secret Rabbit Code' for your resampling. Yes, I've just been working on a resampler recently. Its changing a lot, but it's now usable. I'd actually be quite happy to have some feedback on it. Jean-Marc > Hope this

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi Josh (and everyone else on the list), Here's a quick fix for libFLAC.m4 and libFLAC++.m4 for a problem that crops up if you use the macros in an environment where LD_LIBRARY_PATH is set. The macros save LD_LIBRARY_PATH as ac_save_LDPATH, but restore it from ac_save_LD_LIBRARY_PATH. This patch changes ac_save_LDPATH to ac_save_LD_LIBRARY_PATH. Without this, LD_LIBRARY_PATH effectively gets

Hello all, I gave this a stab a few months ago but gave up and just set up a dual boot. I'm running low on disk space now and tired of messing with the muliple partitions, so I'm giving it another go to get Guild Wars running in Linux. I'm runing Wine 0.9.3 on Kubuntu 6.06 (Dapper Drake). I'm able to go through the instillation without issue, but when it actually tries to

Hi all, i need help to implement a voicemail scenario. What i am trying to do is the following. user X dials a direct access for user Y voicemail and is asked to enter a number (e.g 12345678) and then leaves a message. Then asterisk sends a notification with attachement. The problem is that i need the number entered (e.g 12345678) in the subject. Is that possible. thx in advance.

Hi All, I have an NT 4 domain with multiple samba servers. One of my samba fileservers stopped allowing domain login requests. While it can enumerate the domain users with wbinfo -u, and the domain groups with wbinfo -g, getent passwd does not list the domain users. All the other servers in the domain are fine. Any suggestions for how to track down this error? Sincerely, Donald

My FLAC files reside on my Infrant NV (XRAID) and are also backed up on my PC and on an external USB HDD. That makes 3 copies. I've read most everything on the FLAC site, but as I have mentioned I don't know much about structuring a command line. My PC is also quite old (early P4 from 2000 w/256MB RAM) so I think it will take a LOT of time for this to run through over 1200 CDs worth of

Hi Randy, One thing I would note is that speex is designed for 8kHz, 16kHz or 32kHz. 160 samples is equal to 20ms of 8kHz audio. Have you tried resampling from 11.025kHz to 8kHz and then using the speex 8kHz mode? (nb, or narrow band). Or, if you want to preserve the higher quality of your 11.025kHz sample rate, resample to 16kHz and use the wideband speex encoder. I believe there is new

Hi Jean-Marc, Just some initial feedback, I've just tried to build svn head using the Visual Studio 2005 compiler, and had the following issues: 1. Missing definition of M_PI if it's undefined (and it is on this platform) 2. In speex_resampler_process_int, the compliler can't determine the value of *in_len and *out_len at compile time and thus determine the size of the x and y

hi is compression of metadata of the wav file enabled by default or only when using explicitly the new command line option in v1.2.1? thx

I'm using Asterisk 1.4 and I'm wanting to set an account code for incoming calls. In the extensions.conf file I have the following: exten => s,1,SetAccount(1234) exten => s,n,Dial(SIP/1234) Then when I dial the extension the following error message pops up in the CLI: [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 pbx_extension_helper: No application 'SetAccount' for

Hi: I want to use asterisk1.2 but I don't know which version of asterisk1.2 and zaptel1.2 is best.Please offer me one version of asterisk and zaptel and libpri.How about asterisk1.2.24 and zaptel1.2.20.1 and libpri1.2.5?And do they work togather well? Best regards. --------------------------------- Pinpoint customers who are looking for what you sell. -------------- next part

I need to issue some system commands via the Asterisk manager API. From the CLI the ! (system command) works fine, but when connected via the manager API it fails. Does anyone know why, or of a work around? Thanks Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071010/a3ee4b5d/attachment.htm

FLAC 1.1.4 is released; the Sourceforge shell server is down so I can't update the web page, but downloads are here: http://sourceforge.net/project/showfiles.php?group_id=13478 The windows installer for 1.1.4 is not ready yet but should be along soon. If needed you can replace the binaries from the current (1.1.3) installer with the ones in the 1.1.4 .zip on sourceforge. The main

Dear Tim; What is folks? Where I can find it about VPN solution? Regards Bilal > Hi, > > Greetings to All, > > Im looking for some help on configuring VPN on the Asterisk PBX that I > have hosted in US. Im currently in Middle East and as everyone knows > some countries here has taboo to VOIP. Im not able to get phy phones > registered to my PBX as they are blocking SIP

Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")

Just curious as to the functionality of puppet. Does Puppet ensure that a service is up and running as long as puppet is running? Ie, I want to make sure ssh is always running, if for some reason ssh get''s shut down, does puppet start it back up when it does it''s config sync run? Thanks! --------------------------------- Pinpoint customers who are looking for

I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call

there something fishy going on, have about 21 flac files, in a folder, and 9 of them still showing meta tag data in juk. When running metaflac --list, show nothing on all 21 files, however in amarok and juk I can still see some info on 9 flac files. Howdoes one fix these flac files? are they corrupted? flac -t came back ok on all Anyone ? Richard