search for: check_headers

./configure --bundled-libraries && make ............ [3638/3880] Linking default/source4/lib/policy/py-policy.so [3639/3880] Linking default/source4/auth/ntlm/libauth4.so [3640/3880] Linking default/source4/ntvfs/libntvfs.so default/source4/ntvfs/sysdep/inotify_1.o: In function `inotify_setup': inotify.c:(.text+0x503): undefined reference to `inotify_init'

hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]:

I get the following error when trying to play wav files for my IVR menu. Does anyone know what this means or how to fix it? [Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt Thanks! David

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

...omeone please explain this. I also noticed in lib/util/wscript_configure: if Options.options.enable_lttng != False: conf.check_cfg(package='lttng-ust', args='--cflags --libs', msg='Checking for lttng-ust', uselib_store="LTTNG-UST") conf.CHECK_HEADERS('lttng/tracef.h', lib='lttng-st') conf.CHECK_LIB('lttng-ust', shlib=True) if (conf.CONFIG_SET('HAVE_LTTNG_TRACEF_H') and conf.CONFIG_SET('HAVE_LTTNG_UST')): conf.DEFINE('HAVE_LTTNG_TRACEF', '1') conf.env['HAVE_LTTNG_TRACEF...

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

I downloaded recordPad and recorded a wav file and tried playback on asterisk got the same error as before -- WARNING [1225991360] Format.wav.c:132 check_header:unexpected header size 18-- when I recorded in gsm format on my laptop asterisk did playback well I used sox to resample the recorded wav file on the asterisk machine into wav again and asterisk playback worked well. The sound

Following is a log from an attempt to record and playback a file in wav49 format from an AGI script. COMMAND: stream file aa/after_the_tone "" 0 RESULT_LINE: 200 result=0 endpos=41920 RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')} COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

This is really doing my head in! For some reason, my asterisk box can't playback beep.wav. I have this extension defined in my internal context: '10001' => 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Record(/tmp/asterisk/10001:gsm) [pbx_config]

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

...0572.html> I''m trying to experiment with respond_to in order to not repeat myself and create atom feeds out of a "browse" view. In application controller I created a before filter that checks for the extension of the current url and changes the accept header: [code] def check_headers @headers["Accept"] = "application/rss+xml; application/atom+xml" unless request.path.index(/.*\.xml$/).nil? end [/code] Then, in the appropriate controller, I use respond_to like so (it''s test code): [code] respond_to do |wants| wants.html {render :t...

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

I am having trouble compiling ffmpeg to support speex, which didn't work with the ubuntu libspeex-dev package, but looks like it might with the Speex version 1.2rc1 tarball from http://speex.org/downloads/ How do I tell ffmpeg's configure and/or make to use the 1.2rc1 version of libspeex in /usr/local/include instead of the older debian/ubuntu libspeex-dev package in /usr/include/speex?

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

This is resolved: apt-get remove libspeex-dev cd ~/src/speex-1.2rc1/ ./configure --prefix=/usr make; make install cd ../ffmpeg ./configure --enable-libspeex make; make install worked; then I was able to decode a Speex .flv file: ~/flvs$ ffmpeg -i SpeexQ6R16Efalse.flv foo.wav FFmpeg version SVN-r17174, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-libspeex

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

Hi, These patches contain support for some features that are already in Syslinux 4 but weren't working properly on the elflink branch. It's another step closer to feature parity with Syslinux 4. Having to jump through the comboot API for localboot support is less than ideal and I'll eventually fix that, probably when we move a big chunk of code from asm to C. Also, there's a