similar to: Learn Vectorization (Vectorize)

Hi all, I am currently using the code snippet below to minimize a function which I wasn't able to minimize properly with optim (cliffy surface, constraints, poles). Obviously this iteration is probably the poorest way of implementing such a function and it is expectedly slow. I've already written some vectorized functions, but I'm afraid that I'm missing the "big

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

Dear User, I got the following error running a regression quantile: > rq1<-rq(dep ~ ., model=TRUE, data=exo, tau=0.5 ); > summary(rq1) Erro em rq.fit.fnb(x, y, tau = tau + h) : Error info = 75 in stepy: singular design Any hint about the problem? Thanks a lot, ________________________________________ Ricardo Gon?alves Silva, M. Sc. Apoio aos Processos de Modelagem Matem?tica

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

Hello, I was wondering if anyone can think of a straightforward way (without loops) to do the following shuffling: Let's imagine a vector: c(1,1,1,2,2,3,3,3) I would like to derive shuffled vectors __where the same digits are never separated__, although they can be at both ends (periodicity). So the following shuffled vectors are possible: c(2,2,1,1,1,3,3,3) c(2,1,1,1,3,3,3,2)

Hi Here is another patch regarding time in ogg123 which is more controvercial than the other one I send some days ago. (see attachment) When working with Daisy/SMIL [1] files it would be helpfull if the user could stop play at a specific time. Currently .ogg files are not allowed in the Daisy format but that will hopefully happen one day if I work hard at it. A clip in a Daisy file could look

Following is a log from an attempt to record and playback a file in wav49 format from an AGI script. COMMAND: stream file aa/after_the_tone "" 0 RESULT_LINE: 200 result=0 endpos=41920 RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')} COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0

I found imap core in coredump dir: (gdb) bt #0 0x00000000004abdea in message_parse_header_next (ctx=0x79c150, hdr_r=0x7fffffffe478) at message-header-parser.c:114 #1 0x00000000004a8193 in read_header (mstream=0x795330) at istream-header-filter.c:162 #2 0x00000000004a84e7 in i_stream_header_filter_read (stream=0x795330) at istream-header-filter.c:288 #3 0x00000000004bc216 in i_stream_read

Hi all, I noticed the following Subversion commit today: r6719 | xiphmont | 2004-05-18 16:04:53 +1000 (Tue, 18 May 2004) | 11 lines Updated doc to reflect current proposal... Not as much a proposal at this point actually; this is the way I'm now implementing it. Although we're still in the 'RFC'/'look for horrible lossage' stage, this is close to being

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and have run into some very strange behaviour and I think this is a bug, but I am not completely sure.

Hello, I work at SmarctJog.com, we have here some patches on IceCast for performance and reliability, these are mostly client/connection/source cleanups (a slave merge is underway, and some more good stuff (c)), but we'd like this to be merged in before the list gets any longer. Please find attached a list of our patches with a short desc: This one is actually not from us/me, it was found

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

Hi all, Is it possible to somehow 'bookmark' a place in a sound file? That is, the user presses a key while a sound file is playing and that point is saved, and some time in the future we can play the same sound file and tell it to start playing from that point. This would be done within a perl AGI program. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US

Hi All, When I call the stream_file function all goes well if the user doesn't clear the call. But if I do clear the call (on the handset for example), I get the following exception: -- Channel 0/31, span 1 got hangup RESULT_LINE: 200 result=-1 endpos=28000 == Spawn extension (default, 600006, 1) exited non-zero on 'Zap/31-1'

Hi all, I have a very simple AGI program in Java, but it "sometimes" hangup before the whole process is completed. Looking at the /var/log/asterisk/full, I found the following: AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Rx << STREAM FILE beep "" Jan 20 07:32:30 WARNING[-169026640]: Wait failed (No such file or directory) Jan