similar to: u-law file header ?

Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl -------------- next part -------------- An HTML attachment was scrubbed... URL:

In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'. "Use the NEW ulaw/alaw codec's (slower, but cleaner)" By slower does this mean more 'expensive', or does it instead mean that there will be more algorithmic latency? Both? Can anyone speak to the relative increases? With regard to accuracy, can anyone speak to what kind of situation might

Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via

Does anyone have a Dial script or a hint on how I can dial 10000 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me.

Is there a digium twitter 'user' to follow that only tweets important announcements and release information? If there is not, I think there should be. It would be highly utilitarian to get an SMS when there is an update to Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time with asteriskpbx tweets like: "Anyone trying anything cool with Asterisk over the

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

Does anyone know the mechanism by which companies like YouMail (and MNO's using their own voicemail system) are able to redirect ALL calls from a ALL subscribers to *just one* voicemail DID, yet determine WHICH subscriber did the redirection? I had always assumed this was simply done using RDNIS. In other words, the original calling party's CallerID is passed with the redirected

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Is there a way to use full information maximum likelihood (FIML) to estimate missing data in the sem package? For example, suppose I have a dataset with complete information on X1-X3, but missing data (MAR) on X4. Is there a way to use FIML in this case? I know lavaan and openmx allow you to do it, but I couldn't find anything in the documentation for the sem package. Thanks! -- Dustin Fife

Hi folks. The jspeex library has classes for converting speex to pcm and vice-versa. I also have other code that converts from G.711 to pcm (and vice-versa). I want to transcode G.711 to speex, using an input stream. Can I accomplish this in one step, or must I go G.711 -> PCM -> Speex? If possible in one step, is there some example code I could look at for reference? Thanks! -- Jeff

The ticket clinic, What Is U.S. Copyright Law? The US Copyright Law grants rights to individuals for the works they create. The US Copyright Act of 1790 has changed over the years. The current basis of US copyright law is based on the Copyright Act of 1976. US copyright law is relatively automatic. Once someone has an idea and produces it in tangible form, the creator is the copyright holder and

Let me start with an example: par(mfrow=c(2,3)) for (i in 1:5){ x = rnorm(100) y = .5*x + rnorm(100, 0, sqrt(1-.5^2)) plot(x,y) } Note that there's five plots and six spaces for those plots via mfrow, leaving one row empty. Is there a way to have the bottom two plots centered? I think it looks weird to have them left-justified. Thanks in advance for the help! -- Dustin Fife PhD

Forgive me if this is a trivial question, but I couldn't find it an answer in former forums. I'm trying to reproduce some SAS results where they set two parameters equal. For example: y = b1X1 + b2X2 + b1X3 Notice that the variables X1 and X3 both have the same slope and the intercept has been removed. How do I get an estimate of this regression model? I know how to remove the intercept

I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably earlier versions too) remove the group write permissions from /etc/asterisk/. which is different than 1.4. And 1.6. Is this expected behavior? If so, what's the rationale? If not, I'll submit a bug report if someone hasn't beaten me to it. -K -------------- next part -------------- An HTML attachment was

Hello, As this little program illustrates, duplicating a STRSXP does not seem deep enough. require( inline ) fx <- cfunction( signature( x = "character"), ' SEXP y = PROTECT( duplicate( x ) ); int n = LENGTH(x); int nc = 0 ; char* p = 0 ; for( int i=0; i<n; i++){ p = (char*)( CHAR( STRING_ELT( y , i ) ) ); nc = strlen( p ) ; for( int j=0; j<nc; j++){ p[j] =

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't

Does anyone know enough about the implementation of AstDB to know whether the data structure is a Hash function, a Balanced-Tree, a b-Tree, or a Linked List? I'm trying to estimate the lookup 'cost' of a AstDB with around 160,000 keys? Obviously I already know that it WILL WORK, but the question is whether the data structure is optimal in the Berkeley DB AS IMPLEMENTED in Asterisk.

I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works perfectly, however my theory is that it should be broken. Obviously I'm wrong but "Sip show subscriptions" does not show the endpoint subscribing to the MWI status on Asterisk, even though all of the other endpoints on the system DO subscribe for their respective mailboxes, including SNOM & Polycom endpoints.

I use a script to get the lastest samba: #!/usr/bin/ksh echo "\npassword: cvs\n" cvs -d :pserver:cvs@pserver.samba.org:/cvsroot login cvs -z5 -d :pserver:cvs@pserver.samba.org:/cvsroot co samba Are there any other commands I could tack on or something I should do differently to make sure I am getting correct rebuilds? For example, should I tack on at the end: cd