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...flac --best -o %o --tag=Artist=%{artist} --tag=Album=%{albumtitle} --tag=Date=%{year} --tag=Title=%{title} --tag=Tracknumber=%{number} --tag=Genre=%{genre} %f Recently I've had some problems with burning files with K3B, so I tried this command to decode one flac file to a wav file: flac -d amj.flac And I got the following output: flac 1.1.2, Copyright (C) 2000,2001,2002,2003,2004,2005 Josh Coalson flac comes with ABSOLUTELY NO WARRANTY. This is free software, and you are welcome to redistribute it under certain conditions. Type `flac' for details. amj.flac: 97% complete amj....

...o of the files that were not working. I've uploaded them to my website, but I won't be leaving them there long (only a few days), so if a flac programmer or two can tell me when they're downloaded so I can remove them, I'd appreciate it. Here are the links: http://halblog.com/amj.flac http://halblog.com/CCC.flac As you can see from above, I'm using flac 1.1.2. This is on Ubuntu Feisty Fawn, which, I'm sure everyone knows, is the latest release. This problem may be fixed in later versions, but I would hope a backport and security update is possible, since Ubunt...

Dear all, I have a problem with the function Julian, may be a bug in the function ? Here is a vector of character, which represents dates (May 18 to May 20 2000): > amj <- c("2000-05-18","2000-05-18","2000-05-18","2000-05-19","2000-05-19" > ,"2000-05-19", "2000-05-19", "2000-05-20", "2000-05-20", "2000-05-20") I load the date and chron libraries, I define th...

On Thursday 01 February 2007 20:32, you wrote: > --- Hal Vaughan <hal@thresholddigital.com> wrote: > > amj.flac: 97% complete > > > > amj.flac: ERROR while decoding data > > state = FLAC__STREAM_DECODER_END_OF_STREAM > > > > The file plays without any problem from Amarok. How can flac > > encode this without any problem and have problems decoding it? > &...

...he call. The way around this is to have your sip devices register under different accounts and create a ring group (dial(SIP/dev1&SIP/dev2&SIP/devN)) AFAIK, there isn't a reliable method of determining if a sip device is busy other than calling it. On 5/1/06, Arne Morten Johansen <amj@avio.no> wrote: Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions t...

Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user

[outgoing-fax] exten => _0XXXXXXXXX,1,SetVar(NumberCalled=${EXTEN}) exten => _0XXXXXXXXX,2,Wait(10) exten => fax,1,SetCallerid(${FAX_CALLERID}) exten => fax,2,Dial(Zap/g1/${NumberCalled},60) exten => fax,3,Hangup exten => t,1,Busy exten => i,1,Busy -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

Hello. I just emerged spanDSP with all the packages needed. After a bit of configuration i was read to test. But i get this errormessage stating that application rxfax was not found. I could't fint rxfax i teh modules directory. I use asterisk 1.0.7. I did reset the server after emerging SpanDSP I use gentoo kernel 2.6 I don't know what else to do. Regards, Arne Morten

Hi. I'm having trouble controlling the user info when sending e-mails from asterisk via sendmail to a Microsoft exchange server. When I receive the email the sender is always "asterisk@TheDomainISpecify.com" and the name of the sender is always "Added by portage for asterisk". I want to change both sender-address and the name of the sender. I'm using Gentoo for my

...es@lists.digium.com] P? vegne av Tomislav Parcina Sendt: 22. februar 2006 08:52 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail In article <9ECBC164018FDC4FB48545595C0FB5132D3310@trd-sbs1.AVIO.local>, amj@avio.no says... > Hi. I'm having trouble controlling the user info when sending e-mails > from asterisk via sendmail to a Microsoft exchange server. > > When I receive the email the sender is always > "asterisk@TheDomainISpecify.com" and the name of the sender is alway...

Oh. So how can I do this? If I write something in PHP, how do I make it output to an Asterisk variabel? I need to set a variable in asterisk to TRUE or FALSE based on the result of the PHP-script. ________________________________ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Morgan Gilroy Sendt: 8. februar 2006 15:28 Til:

Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions to this list. I'm looking for a way to make the other phones in a group unavailable when one of them is busy. Because

I also want to try that asterisk guide. But i'm not sure if i understood it correctly. What exactly do i need to do? Do i need to compile Asterisk with the spanDSP plugin or just configure extensions.conf? The URL to spanDSP in the guide wasn't working. I also use a traditional internet line to recieve calls and hopefully i will get Fax working soon. This is so confusing. Thanks, Arne

What about faxing yourself if you don't have a scanner? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Johan van Tongeren Sendt: 5. september 2005 09:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] sending fax [macro-fax-dialing] exten =>

Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail(${mailboxnr}@context) Thanks.

Hello - I would like to use the ruby debugger instead of irb for a breakpoint enabled by script/breakpointer in order to step through some code. How do I do that? Aaron

Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The

Hi there. Is it possible to have different sip users have the same CallerId number in sip.conf. I need this because we got multiple companies on this Asterisk box. Company A's internal numbers: CID: User: 1000 - User 1 2000 - User 2 3000 - User 3 4000 - User 4 Company B's internal numbers: CID: User: 1000 - User 5 2000 - User 6 3000 - User 7 4000 - User 8 Is this allowed? Regards

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