[asterisk-users] Audio Files
Hi people, What can I use to transfer the audio files to and from Asterisk? I was searching and I found the following commands: PUT SOUNDFILE and GET SOUNDFILE They are new commands of AGI, but is there another way to do that? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Files
On Fri, 18 Sep 2009, Anahi Ludue?a wrote: What can I use to transfer the audio files to and from Asterisk? I was searching and I found the following commands: PUT SOUNDFILE and GET SOUNDFILE They are new commands of AGI, but is there another way to do that? From a shell command line, I use rsync, scp, mv, or cp. What are you trying to accomplish? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio Files
Hello there, I wan to know what is the files that have the control of the quality the sound, When I call a extension, and reproduced a file gsm, or I tolk why another extension, have noise... I thinks that is because have bad quality in the .conf. Thanks. Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Files
Hi! I think all - at least all PSTN - calls have the same quality in means of bitrate, number of channels and samplerate. It's 8kHz, 16bit and mono. About noise, I didn't have problems with that. Seems it's not really about quality. Probably it would be helpful, if you tell us, which extensions/protocol you used. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Files
- Original Message - From: Julien Claassen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 26, 2008 8:03 PM Subject: Re: [asterisk-users] Audio Files Hi! I think all - at least all PSTN - calls have the same quality in means of bitrate, number of channels and samplerate. It's 8kHz, 16bit and mono. About noise, I didn't have problems with that. Seems it's not really about quality. Probably it would be helpful, if you tell us, which extensions/protocol you used. Kindest regards Julien Well, I had installed the sample with gmake, and I add my own extension, exten = 269544,1,dial(Sip/user1,20) exten = 269544,2,hangup() and exten = 269544,1,dial(Sip/user2,20) exten = 269544,2,hangup() exten = 1,1,Playback(Wellcome) exten = 1,2,hangup() So, When I call from user1 to user2, have noise, If I call from user1/user2 to extension 1 the Playback have noise to. but, If I call to inexitent extension like the asterisk reproduced a error sound and not have noise.. What's is wrong?? Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio Files, Filtering, and Formats for Asterisk
I listened to all the demos you showed. My ear discerns a little muffling and minor slushiness in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to de-ess the recording. Or, I could just be rather tired. Either way, your best bet is to have the system use WAV files but also make available GSM and ulaw versions of the same files. This is so that the system can pick (and it does this automatically) the best format that requires the least amount of CPU power for transcoding. If I remember correctly, FXO/FXS cards can use straight PCM files. What I do to prepare a file is this: I record in 48Khz 32 bit mode (32 bit is just higher resolution so processing has more to work with, 16 bit is fine since it's the end result), then I compress with de-essing and pop removal (maximizes volume, removes slushy or whistling s's and popping p's), then run through an FFT (Fast Fourier Transform) to bring the frequency response within the natural range of a telephone (if I remember correctly it's within the 300 - 4K range, I can check if need be). Then I reduce the file to a 8Khz, 16 bit mono file. Then I check the audio again, and normalize (like compression but only raises the whole file to where the highest peak of audio reaches the level requested, instead of raising or lowering the level on a dynamic basis by using readahead of a couple milliseconds). I usually normalize to around -3db. The end result is a WAV file that sounds good over the phone. I then put the file on the asterisk server (or another server with sox installed) and convert to gsm, ulaw, and alaw (using the original WAV, not using converted gsm or whatever). Hope this was helpful, and I wish you luck. If nothing else, for like $50 or so (depending on how many files there are) I would be willing to take a series of WAV files from you and perform the filtering and whatnot for you, and supply gsm, wav, ulaw, and alaw. I'm also going to just try and get a series of recordings together through my partner's studio in Phoenix AZ, make the files available to the community, and take up donations for the studio and processing time. Talk to you soon. I'm going to cc the asterisk-users list for this, so that the community can benefit from the information. SKM --Original Message- -From: Stephen Bosch [mailto:[EMAIL PROTECTED] -Sent: Thursday, September 29, 2005 12:18 AM -To: Sherwood McGowan - -Hi, Sherwood: - -If you'll forgive me, I'd like to e-mail you directly with a -few comments and questions. - -Sherwood McGowan wrote: - I have to barge in here... - - Guys, the reason the audio sounds like hell is most probably for - mulitple reasons. First, what codec is the prompt encoded? -Next, what - codec is the client using? If you're using, for example SIP clients - with G711u, asterisk has to re-encode (on the fly I might add) the - prompts to ulaw format to be usable to that client. Only on -GSM based - clients would the audio sound mildly like the actual file. Also, - what's the current load, memory, how many calls are running on the - server, how many calls on hold listening to that music? If -you're not - encoding only one format for those calls, guess what, -you're incurring load to re-encode to each format in use on the fly. - -It's obvious I have much to learn about file formats, but -bear with me here. - -In this particular case, the card is the Digium TDM-400 and -the phone is an analog phone (an old ITT touch-tone phone -- -indestructible and superb sound quality). I don't know if -there's any transcoding happening on-the-fly. - -The thing is, though -- the prompts don't sound that great -when I play them with play on my workstation -- and it's -using the libgsm library, so there shouldn't be any -transcoding happening there. - -All this time I've just been talking about the prompts -provided with Asterisk. Asterisk records voicemail (for -example) in the same GSM format that these prompts are in (if -the recordings sound as poor as these prompts do, I don't -know whether I'll even be able to use Asterisk for this application). - -While the delivery is professional and I'm sure the original -sources sound great, these GSM files don't sound so hot. -They're muffled and there is a slight bit of static. I've -attached the demo-congrats.gsm file for example. Try it yourself. - - I am a music producer, have been for several years. One of -the things - I do on the side from my day job as a VOIP Admin/Engineer is make - prompts and music for customer's PBXs. I typically make a WAV file, - and then do my filtering, compressing, and finally -normalizing, then - save it. Finally, I put the files in the server in question and use - sox to re-encode multiple - formats: gsm, ulaw, alaw, etc You'll find the results are much - better than just
[Asterisk-Users] Audio files problem - as usual
Hello List! I have a problem that has been posted to the list more than once, but so far I have not been able to find a solution searching the archives and Google. The problem is with Asterisk audio files not being played to the x-lite client. I have an out-of-the-box [EMAIL PROTECTED] configuration with no additional hardware. I have created extensions, clients over the LAN are able to talk to each other and I can even listen to the MP3 files that come out of the box with the mp3play command (both on the Linux box and on the phone). But all the standard audio files (voicemail, text-to-speech and even music-on-hold) can't be heard. It looks like Asterisk just hangs until I hang up. Sounds familiar? Can you please help me out with this? Thanks, Luca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio Files from a Database
Is there a way to stream or at least load into a variable with AGI, gsm or wav files out of a MySql database (contained in MySql as blob fields) directly from asterisk without having to write the files to disk first before you stream them out? I've seen a hack for mpg123 that lets you open MP3's from a database, but nothing for anything else. Seems like a way to make * a little more dynamic. Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
