similar to: Speex and RTP

> 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 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: 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

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

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

Dear All, We are trying to install CDR Stats module. We are able to open its web pages but unable to retrieve data from CDRs. Can anybody suggest that how to connect this module with Asterisk CDRs database ? Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN:

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

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

Folks: The single reference to a Mac version of flac software that I can find seems to be way out of date. In addition, when I DL the file and attempt to install, I am greeting with an impressive list of "missing" software items that I believed, incorrectly, it seems, was suppose to be included in the file that I did DL. Any suggestions?? BTW, I run Mac OS X 10.4.8. Avi P.

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

committing won''t push to production -- you have tested I assume locally? localhost:3000 shows your changes? if so, I can simply go on to the server where it is deployed and issue: svn export (or svn update) and restart the application for changes to actually propagate. svn log shows your commit, that is all well and good: SeeBQ-MacBook:~/rails_apps/atlrug.org cbq$ svn update U

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

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

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 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