[asterisk-users] Asterisk does not translate from wav to alaw
Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/vprompts/*zip-code.wav* (format 0x8 (*alaw*)): No such file or directory [Aug 28 11:16:29] WARNING[2705]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/test1-000f for /var/lib/asterisk/sounds/vprompts/*zip-code.wav* Am I missing a module to translate from wav to alaw/gsm/g726/... ?? 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] Asterisk does not translate from wav to alaw
On Saturday 28 August 2010 04:22:18 Jonas Kellens wrote: Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/vprompts/*zip-code.wav* (format 0x8 (*alaw*)): No such file or directory [Aug 28 11:16:29] WARNING[2705]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/test1-000f for /var/lib/asterisk/sounds/vprompts/*zip-code.wav* Am I missing a module to translate from wav to alaw/gsm/g726/... ?? Please drop the .wav extension from the Playback/Background. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Asterisk does not translate from wav to alaw
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens jonas.kell...@telenet.bewrote: Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/vprompts/*zip-code.wav* (format 0x8 (*alaw*)): No such file or directory [Aug 28 11:16:29] WARNING[2705]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/test1-000f for /var/lib/asterisk/sounds/vprompts/*zip-code.wav* Am I missing a module to translate from wav to alaw/gsm/g726/... ?? My guess is that your .wav file is NOT 8khz. The system doesn't accept anything but wav files at 8khz. Use sox to downsample to 8khz (and 1 chan), and the problems should go away. While you are at it, you could use sox to convert to the target format in a single operation. The scripts that Digium uses to take Allison's voice prompts (at 48khz) to the different formats, convert things to slin (raw) as a central format, but in my experience, the fewer steps the better. But I doubt that anyone could detect the difference in the end result... Here's what I do with CD-qual sounds to turn them into the common Asterisk formats: Assume $i is the name of the .wav file you want to convert: x=`basename $i .wav` sox -v 0.7 $i -r 16000 -c 1 -t sw $x.sln16 sox -v 0.7 $i -r 8000 -c 1 -t sw $x.raw sox -r 8000 -c 1 -t sw $x.raw -t gsm $x.gsm## OR ### sox -v 0.7 $i -r 8000 -t gsm $x.gsm sox -r 8000 -c 1 -t sw $x.raw -t ul $x.ulaw## OR ### sox -v 0.7 $i -r 8000 -t ul $x.ulaw sox -r 8000 -c 1 -t sw $x.raw -t al $x.alaw ## OR ### sox -v 0.7 $i -r 8000 -t wav $x.wav rm $x.raw y=`pwd` sudo asterisk -rx file convert $y/$i $y/$x.g722 I'm ignoring the siren g729 formats; use asterisk for those in like format, depending on your asterisk version and codecs. Allison normalizes the volume of sounds she distributes; use the -v 0.7 to bring the volume down a bit to the standard, and your sounds won't stick out against rest of Allison's existing recordings in Asterisk. Digium uses a filter program to 'heighten' the sounds a little; That's the main reason, I think, that they use the .raw format as an in-between. I've been skipping this step, as it doesn't seem critical, in which case the direct conversion is probably preferable. I suggest, that if you are converting sounds for Asterisk's sake, that you convert to all the possible formats. Disk space is cheap, and you'll squeeze a little extra performance out of Asterisk by allowing it to pick the 'best' format. Dahdi type interfaces would prefer the ulaw/alaw formats; High-def phones like Snom (and appropriate Polycoms, etc) could use the g722. Ulaw and gsm transcodings are cheap, but no transcoding is cheaper still. murf 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 -- Steve Murphy ParseTree Corp -- _ -- 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] Asterisk does not translate from wav to alaw
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/zip-code.wav does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/vprompts/zip-code.wav (format 0x8 (alaw)): No such file or directory [Aug 28 11:16:29] WARNING[2705]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/test1-000f for /var/lib/asterisk/sounds/vprompts/zip-code.wav Am I missing a module to translate from wav to alaw/gsm/g726/... ?? My guess is that your .wav file is NOT 8khz. The system doesn't accept anything but wav files at 8khz. Use sox to downsample to 8khz (and 1 chan), and the problems should go away. While you are at it, you could use sox to convert to the target format in a single operation. The scripts that Digium uses to take Allison's voice prompts (at 48khz) to the different formats, convert things to slin (raw) as a central format, but in my experience, the fewer steps the better. But I doubt that anyone could detect the difference in the end result... Here's what I do with CD-qual sounds to turn them into the common Asterisk formats: Assume $i is the name of the .wav file you want to convert: x=`basename $i .wav` sox -v 0.7 $i -r 16000 -c 1 -t sw $x.sln16 sox -v 0.7 $i -r 8000 -c 1 -t sw $x.raw sox -r 8000 -c 1 -t sw $x.raw -t gsm $x.gsm## OR ### sox -v 0.7 $i -r 8000 -t gsm $x.gsm sox -r 8000 -c 1 -t sw $x.raw -t ul $x.ulaw## OR ### sox -v 0.7 $i -r 8000 -t ul $x.ulaw sox -r 8000 -c 1 -t sw $x.raw -t al $x.alaw ## OR ### sox -v 0.7 $i -r 8000 -t wav $x.wav rm $x.raw y=`pwd` sudo asterisk -rx file convert $y/$i $y/$x.g722 I'm ignoring the siren g729 formats; use asterisk for those in like format, depending on your asterisk version and codecs. Allison normalizes the volume of sounds she distributes; use the -v 0.7 to bring the volume down a bit to the standard, and your sounds won't stick out against rest of Allison's existing recordings in Asterisk. Digium uses a filter program to 'heighten' the sounds a little; That's the main reason, I think, that they use the .raw format as an in-between. I've been skipping this step, as it doesn't seem critical, in which case the direct conversion is probably preferable. I suggest, that if you are converting sounds for Asterisk's sake, that you convert to all the possible formats. Disk space is cheap, and you'll squeeze a little extra performance out of Asterisk by allowing it to pick the 'best' format. Dahdi type interfaces would prefer the ulaw/alaw formats; High-def phones like Snom (and appropriate Polycoms, etc) could use the g722. Ulaw and gsm transcodings are cheap, but no transcoding is cheaper still. murf Steve Thanks for sharing I appericate your insight as this is something I run up against as well. What about g729 we use this coded a lot what is the best method to transcode it it? Thanks Bryant -- _ -- 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] Asterisk does not translate from wav to alaw
On Saturday 28 August 2010 10:35:59 Bryant Zimmerman wrote: Thanks for sharing I appericate your insight as this is something I run up against as well. What about g729 we use this coded a lot what is the best method to transcode it it? If you load res_convert.so, you will have a CLI command file convert Usage: file convert file_in file_out Convert from file_in to file_out. If an absolute path is not given, the default Asterisk sounds directory will be used. Example: file convert tt-weasels.gsm tt-weasels.ulaw -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Asterisk does not translate from wav to alaw
On Sat, Aug 28, 2010 at 9:52 AM, Tilghman Lesher tles...@digium.com wrote: On Saturday 28 August 2010 10:35:59 Bryant Zimmerman wrote: Thanks for sharing I appericate your insight as this is something I run up against as well. What about g729 we use this coded a lot what is the best method to transcode it it? If you load res_convert.so, you will have a CLI command file convert Usage: file convert file_in file_out Convert from file_in to file_out. If an absolute path is not given, the default Asterisk sounds directory will be used. Example: file convert tt-weasels.gsm tt-weasels.ulaw Tilghman speaks rightly. This conversion utility keys from file names, so for g729, you might say: asterisk -rx file convert tt-weasels.ulaw tt-weasels.g729 or, asterisk -rx file convert /home/user/tt-weasels.gsm /home/user/tt-weasels.g729 (oh, and make sure asterisk is running!) murf PS. Wouldn't it be nice if sox could handle the sirens, g729, and everything else? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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
