Re: [asterisk-users] realtime sip peers : musiconhold class
Converting with sox works well as followed : sox -V intro.wav -c 1 -r 8000 intro2.wav To convert with asterisk convert, I needed to use an absolute path : asterisk -rx file convert /var/lib/asterisk/moh/folder/intro.wav /var/lib/asterisk/folder/intro.alaw All works well. Jonas. On 08/19/2010 02:31 AM, Nasir Iqbal wrote: Hi to convert wav file use following sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile' while replace orgFile and fixedFile with actual filenames If still now luck try with mp3 Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Hi to convert wav file use following sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile' while replace orgFile and fixedFile with actual filenames If still now luck try with mp3 Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Can anyone help because I don't understand why Asterisk can not read the input file, there is not much info given... 2 files : [r...@asterisk testing]# file testExtended.wav testExtended.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz [r...@asterisk testing]# file testLong.wav testLong.wav: RIFF (little-endian) data, WAVE audio, 20294 channels 1414676809 Hz to mono : [r...@asterisk testing]# sox testExtended.wav -r 8000 -c1 testExtended2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative [r...@asterisk testing]# sox testLong.wav -r 8000 -c1 testLong2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox effects: resample clipped 2 samples; decrease volume? afterwards : [r...@asterisk testing]# file testLong2.wav testLong2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz [r...@asterisk testing]# file testExtended2.wav testExtended2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz But Asterisk can not open them : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav Any thoughts ?! Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Hi. Just to let you know, we record voices with audacity, and export audio as flac, just in case we need to edit it. Then I have the following sh script: o# cat convert.sh #!/bin/sh today=$(date +%F); mkdir -p $today/flac; mkdir -p $today/wav; mkdir -p $today/ul; for i in *.flac; do echo echo Processing $i; echo #$filename= sox $i -r 8000 -c 1 $(echo $i|rev|cut -d . -f2-10|rev).wav; normalize-audio -a 25dB $(echo $i|rev|cut -d . -f2-10|rev).wav; mv $i $today/flac/; sox $(echo $i|cut -d . -f1).wav $(echo $i|rev|cut -d . -f2-10|rev).ul; mv $(echo $i|rev|cut -d . -f2-10|rev).wav $today/wav/; mv $(echo $i|rev|cut -d . -f2-10|rev).ul $today/ul/; echo ; done echo All done; On 17 August 2010 08:07, Jonas Kellens jonas.kell...@telenet.be wrote: Can anyone help because I don't understand why Asterisk can not read the input file, there is not much info given... 2 files : [r...@asterisk testing]# file testExtended.wav testExtended.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz [r...@asterisk testing]# file testLong.wav testLong.wav: RIFF (little-endian) data, WAVE audio, 20294 channels 1414676809 Hz to mono : [r...@asterisk testing]# sox testExtended.wav -r 8000 -c1 testExtended2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative [r...@asterisk testing]# sox testLong.wav -r 8000 -c1 testLong2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox effects: resample clipped 2 samples; decrease volume? afterwards : [r...@asterisk testing]# file testLong2.wav testLong2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz [r...@asterisk testing]# file testExtended2.wav testExtended2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz But Asterisk can not open them : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav Any thoughts ?! Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
I took this from the wiki, but it's not working : [r...@asterisk testing]# sox test.wav -r 8000 -c1 test.alaw resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox formats: no handler for file extension `alaw' Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
And even when I think the format is correct : [r...@asterisk testing]# sox test.wav -r 8000 -c1 test.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox wav: Premature EOF on .wav input file [r...@asterisk testing]# file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz It is still not working : [r...@asterisk testing]# asterisk -rx file convert test.wav test.alaw Unable to open input file: test.wav Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
I have another file that reads : [r...@asterisk ]# file intro\ extended\ version.wav intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz With the same result : [r...@asterisk ]# asterisk -rx file convert /var/lib/asterisk/moh/test/intro\ extended\ version.wav /var/lib/asterisk/moh/test/testing.alaw Unable to open input file: /var/lib/asterisk/moh/test/intro extended version.wav Jonas. On 08/13/2010 01:49 PM, Gareth Blades wrote: The wav file is not in the correct format. Also the number of channels and sampling frequency it is reporting is complete nonsense. This is what it should display:- RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime sip peers : musiconhold class
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 following in the column '*musiconhold*' : *106002* asterisk*CLI moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh Class: 106002 Mode: files Directory: /var/lib/asterisk/moh/106002 But always : [Aug 13 09:47:57] -- Started music on hold, class 'default', on SIP/test2-0014 [Aug 13 09:48:05] -- Stopped music on hold on SIP/test2-0014 Can anyone help ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Hello list, when putting the class 'default' in comment, then this happens : [Aug 13 12:36:34] WARNING[21172]: res_musiconhold.c:666 get_mohbyname: Music on Hold class 'default' not found [Aug 13 12:36:34] -- Started music on hold, class '106002', on SIP/test2-0001 [Aug 13 12:36:34] WARNING[21172]: format_wav.c:124 check_header: Does not say fmt [Aug 13 12:36:34] WARNING[21172]: file.c:385 fn_wrapper: Unable to open format wav [Aug 13 12:36:34] WARNING[21172]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/106002/01Long': No such file or directory Questions : 1. how can I use AND class default AND class 106002 ?! 2. is it normal that Asterisk can not convert from wav to alaw/gsm ?! Jonas. On 08/13/2010 09:57 AM, Jonas Kellens wrote: 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 following in the column '*musiconhold*' : *106002* asterisk*CLI moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh Class: 106002 Mode: files Directory: /var/lib/asterisk/moh/106002 But always : [Aug 13 09:47:57] -- Started music on hold, class 'default', on SIP/test2-0014 [Aug 13 09:48:05] -- Stopped music on hold on SIP/test2-0014 Can anyone help ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Asterisk can convert from wav but it still needs to be in the correct format. See http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk Jonas Kellens wrote: Hello list, when putting the class 'default' in comment, then this happens : [Aug 13 12:36:34] WARNING[21172]: res_musiconhold.c:666 get_mohbyname: Music on Hold class 'default' not found [Aug 13 12:36:34] -- Started music on hold, class '106002', on SIP/test2-0001 [Aug 13 12:36:34] WARNING[21172]: format_wav.c:124 check_header: Does not say fmt [Aug 13 12:36:34] WARNING[21172]: file.c:385 fn_wrapper: Unable to open format wav [Aug 13 12:36:34] WARNING[21172]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/106002/01Long': No such file or directory Questions : 1. how can I use AND class default AND class 106002 ?! 2. is it normal that Asterisk can not convert from wav to alaw/gsm ?! Jonas. On 08/13/2010 09:57 AM, Jonas Kellens wrote: 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 following in the column '*musiconhold*' : *106002* asterisk*CLI moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh Class: 106002 Mode: files Directory: /var/lib/asterisk/moh/106002 But always : [Aug 13 09:47:57] -- Started music on hold, class 'default', on SIP/test2-0014 [Aug 13 09:48:05] -- Stopped music on hold on SIP/test2-0014 Can anyone help ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
1. the converting is not working [r...@asterisk testing]# file 01Long.wav 01Long.wav: RIFF (little-endian) data, WAVE audio, 20294 channels 1414676809 Hz [r...@asterisk testing]# asterisk -rx file convert /var/lib/asterisk/testing/01Long.wav /var/lib/asterisk/testing/01Long.alaw Unable to open input file: /var/lib/asterisk/testing/01Long.wav 2. This does not explain why I can't use class default AND class whatever. Jonas. On 08/13/2010 12:47 PM, Gareth Blades wrote: Asterisk can convert from wav but it still needs to be in the correct format. See http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
The wav file is not in the correct format. Also the number of channels and sampling frequency it is reporting is complete nonsense. This is what it should display:- RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Jonas Kellens wrote: 1. the converting is not working [r...@asterisk testing]# file 01Long.wav 01Long.wav: RIFF (little-endian) data, WAVE audio, 20294 channels 1414676809 Hz [r...@asterisk testing]# asterisk -rx file convert /var/lib/asterisk/testing/01Long.wav /var/lib/asterisk/testing/01Long.alaw Unable to open input file: /var/lib/asterisk/testing/01Long.wav 2. This does not explain why I can't use class default AND class whatever. Jonas. On 08/13/2010 12:47 PM, Gareth Blades wrote: Asterisk can convert from wav but it still needs to be in the correct format. See http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
