search for: format_wav

Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:24...

Does anyone know what this means "Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call 7438737dc873850@172.16.0.52 for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask.

...ssage below on the CLI. Dell PE 1600 with Sangoma A200. pbtpbx*CLI> show version Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running Linux on 2007-01-13 18:31:56 UTC Asterisk Queue Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) Feb 5 08:43:00 WARNING[20103]: format_wav.c:247 update_header: Unable to find our position == Parsing '/etc/asterisk/logger.conf': Found -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com -- This message ha...

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]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such...

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 14 08...

...o load => codec_adpcm.so load => codec_alaw.so load => codec_a_mu.sothe number of modules to minimum ? load => codec_g722.so load => codec_g726.so load => codec_gsm.so load => codec_lpc10.so load => codec_ulaw.so load => format_gsm.so load => format_pcm.so load => format_wav.so load => format_wav_gsm.so load => res_agi.so load => res_clioriginate.so load => res_fax.so load => res_musiconhold.so load => res_timing_timerfd.so load => func_callerid.so load => func_cdr.so load => func_channel.so load => func_cut.so load => func_math.so lo...

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 /var/spool/asterisk/voicemail/default/5933/unavail.wav Oct 11 19:57:26 WARNING[6587]: file.c:804 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5933/unavail (fo...

...ll at 312 bytes, containing: Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Event Logger restarted Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Queue Logger restarted Aug 29 23:22:17 VERBOSE[24303] logger.c: Rotated Logs Per SIGXFSZ (Exceeded file size limit) Aug 29 23:22:17 WARNING[24303] format_wav.c: Unable to find our position Stopping/starting A*k seems to reset things ok... Adrian Marsh

I'm still getting a sementation fault with mpg123. I have tried different parameters creating mp3s the last from cd audio ... lame -m s --resample 8000 -q 0 -a --cbr -b 32 and several versions of mpg123. I have always created 8000 hz outputs. I've got other * boxes that don't use moh that have been up for months. This one crashes every couple of days - the verbose output leading to a

...00388000 r-xp 00000000 08:02 723038 /usr/lib/asterisk/modules/res_crypto.so 00388000-00389000 rw-p 00003000 08:02 723038 /usr/lib/asterisk/modules/res_crypto.so 00389000-0038c000 r-xp 00000000 08:02 722956 /usr/lib/asterisk/modules/format_wav.so 0038c000-0038d000 rw-p 00002000 08:02 722956 /usr/lib/asterisk/modules/format_wav.so 0038d000-00390000 r-xp 00000000 08:02 722950 /usr/lib/asterisk/modules/format_pcm.so 00390000-00391000 rw-p 00002000 08...

...uses ultra-wideband iLBC; it's audio quality is also good. But I understand there are patent problems. (Are licenses even available? How much do they cost?) I've got three problems for actually using this though: 1) Can asterisk load 32k audio files? I see that the current CVS's format_wav.c can not. Is there another module that does? I might try modifying the WAV loader. Are there assumptions of 8k throughout Asterisk, or is this pretty isolated to that file? 2) Are there higher quality versions of the Asterisk sounds? They're in a nice, professional voice, but 8K GSM jus...

...ming_timerfd.moduleinfo /usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 1.8.7.0. NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so. Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind? SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did watch -n1...

..., "SIP/344|20|wWtT") in new stack -- Called 344 -- SIP/344-5e4e is ringing -- SIP/344-5e4e answered SIP/341-5931 -- Attempting native bridge of SIP/341-5931 and SIP/344-5e4e -- Started music on hold, class 'native', on SIP/344-5e4e Mar 2 11:17:50 WARNING[7717]: format_wav.c:161 check_header: ot in mono 2 ar 2 11:17:50 WARNING[7717]: file.c:432 ast_filehelper: nable to open file on / var/lib/asterisk/moh-native/fpm-sunshine.wav ar 2 11:17:50 WARNING[7717]: res_musiconhold.c:225 ast_moh_files_next: nable to open file '/var/lib/asterisk/moh-native/fpm-sunshine...

...=> 22,1,MixMonitor(test.wav) exten => 22,2,Dial(SIP/blabla@blabla.com) No problems at all if I record to a file, but then I made test.wav a fifo, and had oggenc read it, then pipe it to oggfwd, etc... This does work, but generates a pile of warnings in the asterisk console: WARNING[16235]: format_wav.c:247 update_header: Unable to find our position I can shut off the warnings, of course, using logger.conf, but it still seems kind of messy. I also need to start the commands reading that pipe manually after initiating the call. Can anyone suggest a better way to do this? I figure an AGI script...

...get the following: -- Executing Answer("IAX2/308-4", "") in new stack -- Executing Wait("IAX2/308-4", "2") in new stack -- Executing Record("IAX2/308-4", "/tmp/asterisk/10001:gsm") in new stack Oct 16 12:49:41 WARNING[8581]: format_wav.c:153 check_header: Not a wav file 49 Oct 16 12:49:41 WARNING[8581]: file.c:436 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/beep.wav Oct 16 12:49:41 WARNING[8581]: file.c:824 ast_streamfile: Unable to open beep (format ulaw): No such file or directory Oct 16 12:49:41 WARNING[...

...ect Coder/Decoder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [format_g723.so] => (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g723 [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav *CLI> show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM...

...ect Coder/Decoder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [format_g723.so] => (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g723 [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav *CLI> show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM S...

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

...und files: root@asterisk1 modules]# cd /usr/lib/asterisk/modules/ root@asterisk1 modules]# ls -1 format_* format_au.so format_g723.so format_g726.so format_g729.so format_gsm.so format_h263.so format_ilbc.so format_jpeg.so format_mp3.so format_pcm_alaw.so format_pcm.so format_sln.so format_vox.so format_wav_gsm.so format_wav.so So I imagine I can use .au and .mp3 format for audio files. That's what I want. But I cannot find now the appropriate format to specify in voicemail.conf format directive. I set format=mp3 first and format=au But when the voicemail prompt for the message I have 0 secon...