search for: format_gsm

...(language 'en') -- Playing 'astcc-accountnum2' (language 'en') Jan 14 16:48:19 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data Jan 14 16:48:19 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data < ...> Jan 14 16:48:20 WARNING[15662]: format_gsm.c:155 gsm_read: Short read (1) (No such file or directory)! -- Playing 'digits/14' (language 'en') Jan 14 16:48:22 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data Jan 14 16:48:22 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data < ...&gt...

...oad => app_record.so load => app_userevent.so 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...

...playback or voicemail doesnt play anything .... but when i dial 102, I hear the MP3 music .. exten => 99,1,Answer() exten => 99,2,Playback(prepaid-welcome) exten => 99,3,Hangup() exten => 101,1,VoiceMailMain() exten => 102,1,Answer() exten => 102,2,MusicOnHold(default) I have format_gsm.so, codec_gsm.so loaded and i am using asterisk-sounds-1.2.1 , asterisk-1.2.23 on debian 4.0. do i miss any audio library? thanks atik

...hone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so and format_gsm.so, recompiled the kernel over and over, but I just can't figure it out. Any one got some ideas? Modem.conf: [interfaces] context=incomming driver=i4l dialtype=tone mode=immediate language=en type=autodetect group=1 msn=39660425 incomingmsn=39660425,39660426 device => /dev/ttyI0 device...

....gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted warnings and errors in the CLI, apparently coinceding with corrupt MixMonitor recordings: format_gsm.c:65 gsm_read: Short read (13) (Resource temporarily unavailable)! WARNING[30727]: app_dial.c:1379 wait_for_answer: Unable to write frametype: 2 WARNING[14712]: file.c:766 ast_readaudio_callback: Failed to write frame WARNING[2612]: file.c:766 ast_readaudio_callback: Failed to write frame WARNI...

...ion 'MusicOnHold' ............... == Registered application 'DateTime' app_sayunixtime.so => (Say time) == Registered application 'SetCallerPres' app_setcallerid.so => (Set CallerID Presentation Application) == Registered file format gsm, extension(s) gsm format_gsm.so => (Raw GSM data) == Registered application 'BackgroundDetect' app_talkdetect.so => (Playback with Talk Detection) Segmentation fault strace was little help: strace asterisk -c ....... .open("/usr/lib/asterisk/modules/format_gsm.so", O_RDONLY) = 12 read(12, "\...

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

...to start with a fresh setup. [app_read.so] => (Read Variable Application) == Registered application 'Read' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver' [format_gsm.so] => (Raw GSM data) == Registered file format gsm, extension(s) gsm [app_dial.so] => (Dialing Application) == Registered application 'Dial' [app_striplsd.so] => (Strip trailing digits) == Registered application 'StripLSD' [app_disa.so] => (DISA (Direct Inward...

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

the problem below keeps recarrying even after i have cleared this error when i run asterisk -vvv or -c the error occurs again please help ..Warning, flexible rate not heavily tested! .................WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File

...ing wav files in my basic linux environement. So my plan is to switch to another codec. I checked about the available modules for sound 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...

...d the clients are?nt able to make a call anymore. Asterisk says: ?rejected connect attempt? The next problem is the port 4569, if I activate the line ?PORT=4569? in the IAX.conf ? Asterisk says: ?Ignoring Port for Now? And sometimes asterisk says: Apr 20 23:46:20 WARNING[789526]: format_gsm.c:142 gsm_read: Short read (23) (Interrupted system call)! Sometimes calls are disconnected ? Both tested with Asterisk 0.7.2 from diginum ftp and with cvs 16.04.04 We use: Asterisk cvs 16.04.04 On debian kernel 2.4.26 Without additional isdn or BRI/PRI...

...= Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder) == Registered translator 'g726tolin' from format g726 to slin, cost 10 == Registered translator 'lintog726' from format slin to g726, cost 10 [format_gsm.so] => (Raw GSM data) == Registered file format gsm, extension(s) gsm [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM)) == Registered file format wav49, ex...

..._sip.so noload => chan_skinny.so load => codec_a_mu.so load => codec_adpcm.so load => codec_alaw.so load => codec_g726.so load => codec_gsm.so load => codec_ilbc.so load => codec_lpc10.so load => codec_ulaw.so load => format_g726.so load => format_g729.so load => format_gsm.so load => format_h263.so load => format_ilbc.so load => format_jpeg.so load => format_pcm.so load => format_pcm_alaw.so load => format_vox.so load => format_wav.so load => format_wav_gsm.so load => cdr_csv.so load => pbx_config.so load => pbx_spool.so load => pb...

...0 app_setcdruserfield.so CDR user field apps 0 app_enumlookup.so ENUM Lookup 0 pbx_config.so Text Extension Configuration 0 app_read.so Read Variable Application 0 format_gsm.so Raw GSM data 0 app_dial.so Dialing Application 0 app_striplsd.so Strip trailing digits 0 app_cut.so Cuts up variables 0 app_echo.so Simple E...

...load => codec_ulaw.so load => codec_alaw.so load => format_pcm.so load => func_uri.so ;required by app_dial and chan_sip load => res_features.so load => app_dial.so ;playback and echo apps ... load => app_playback.so load => app_echo.so load => codec_gsm.so load => format_gsm.so load => format_wav_gsm.so load => chan_h323.so load => chan_sip.so load => chan_local.so When I do: > stop now asterisk hangs up, but locks: *CLI> stop now Beginning asterisk shutdown.... Executing last minute cleanups Asterisk cleanly ending (0). I attached gdb to the l...

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

...ding symbols from /usr/lib/asterisk/modules/codec_a_mu.so...done. Loaded symbols for /usr/lib/asterisk/modules/codec_a_mu.so Reading symbols from /usr/lib/asterisk/modules/codec_g726.so...done. Loaded symbols for /usr/lib/asterisk/modules/codec_g726.so Reading symbols from /usr/lib/asterisk/modules/format_gsm.so...done. Loaded symbols for /usr/lib/asterisk/modules/format_gsm.so Reading symbols from /usr/lib/asterisk/modules/format_wav.so...done. Loaded symbols for /usr/lib/asterisk/modules/format_wav.so Reading symbols from /usr/lib/asterisk/modules/format_wav_gsm.so...done. Loaded symbols for /usr/lib/...

...0 format_pcm.so Raw uLaw 8khz Audio support (PCM) 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 format_wav.so Microsoft WAV format (8000hz Signed Line 0 format_gsm.so Raw GSM data 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 codec_ulaw.so Mu-law Coder/Decoder 0 codec_adpcm.so Adap...

...symbols from /opt/asterisk/lib/modules/format_g726.so...done. Loaded symbols for /opt/asterisk/lib/modules/format_g726.so Reading symbols from /opt/asterisk/lib/modules/format_g729.so...done. Loaded symbols for /opt/asterisk/lib/modules/format_g729.so Reading symbols from /opt/asterisk/lib/modules/format_gsm.so...done. Loaded symbols for /opt/asterisk/lib/modules/format_gsm.so Reading symbols from /opt/asterisk/lib/modules/format_h263.so...done. Loaded symbols for /opt/asterisk/lib/modules/format_h263.so Reading symbols from /opt/asterisk/lib/modules/format_h264.so...done. Loaded symbols for /opt/aster...