similar to: channel name

Hi, when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and start asterisk to be able to use MeetMe(). When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and start asterisk but I am not able to use MeetMe(). What do I miss? Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new

Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Thanks, -- Cheers, Maxim Veksler "Free as in Freedom" - Do u GNU ?

Dear all, I searched the mail archives and the R site and found no guidance (tried "merge", "cbind" and terms like "coalesce" with no success). There surely is a way to coalesce (like in SQL) columns in a dataframe, right? For example, I would like to go from a dataframe with two columns to one with only one as follows: From Name.x Name.y nx1 ny1 nx2 NA

[[this bounced first, because it has 'help' in the Subject line ... -- Martin Maechler ]] I am a complete novice R programmer. (Though I know C quite well) I am trying to write some R code to do the following simulation. There is a 2-frame "movie" of noise and signal dots. the noise dots have random positions in each frame. The signal dots are placed randomly in frame 1,

Hi In the following codes, I had to choose between printing (= TRUE) or deliver something for grab (ei, vi). Is there a way to get both--that is, to print and also have ei and vi for grab? Thanks. Steven ... out<-round(as.data.frame(cbind(ap,se,t,p)),digits) out<-cbind(out,sig) out<-out[!grepl(colnames(zx)[1],rownames(out)),] if(printing){ cat("\nAPPs of bivariate ordered

?s 09:51 de 11/08/2024, Steven Yen escreveu: > Hi > > In the following codes, I had to choose between printing (= TRUE) or > deliver something for grab (ei, vi). Is there a way to get both--that > is, to print and also have ei and vi for grab? Thanks. > > Steven > > ... > > out<-round(as.data.frame(cbind(ap,se,t,p)),digits) > out<-cbind(out,sig) >

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

Thanks. Will try it. Have not tried it but I think the following may work: out$results<-NULL out$results$ei<-ap out$results$vi<-vap All I need is printing by returning out (unless I turn it off). And, retrieve ap and vap as needed as shown above. Guess I need to read more about invisible. On 8/11/2024 10:09 PM, Rui Barradas wrote: > ?s 09:51 de 11/08/2024, Steven Yen escreveu:

Here are another three problems with logarithmic scales: 1) segments() does not work with logarithmic scales. I suggest to change lines 962-973 in "plot.c": for (i = 0; i < n; i++) { if (FINITE(xt(x0[i%nx0])) && FINITE(yt(y0[i%ny0])) && FINITE(xt(x1[i%nx1])) && FINITE(yt(y1[i%ny1]))) { GP->col = INTEGER(col)[i % ncol];

I noticed this in current intermediate patch version: > runif(0,0,4095) numeric(0) > round(runif(0,0,4095)) [1] NA I don't like it, but maybe it is the way it is supposed to be. I would prefer for it to stay numeric(0). I notice also > ceiling(runif(0,0,4095)) numeric(0) > floor(runif(0,0,4095)) numeric(0) which leads me to thing round() is not working right. I have some code

Hi list. We have a problem when dialing over IAX to another Asterisk server: we've setup an extension named 'f19dffb971b93746d73ec46d5f1d4b36c199f48c-g1' in a specific context (its large because it needs to be unique). I've read in past discussions on asterisk-dev list that the extension length is limited to 79 characters - which I though should be more then enough. Now were

Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in).

Hi! I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time now on my Home PC. I want to shift to a PC having SATA hard disk .Can I install Redhat 9.0 on SATA hard disk ??some people are telling me that I have to go for Linux Enterprise 4.0.I don`t want to leave Linux 9.0 because I want to run Asterisk 1.0.3 Can anyone help me?? Amna -------------- next part -------------- An HTML

FYI: I am trying to build zaptel-1.2.4 against the recently updated kernel version 2.6.9-34.EL on Centos 4.2. but I am getting errors and it will not build. This is apparently due to a typo in a kernel header spinlock.h although I have not successfully modified the kernel and built zaptel against it yet. https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 This bug report has a typo

Hey Folks, I''m looking at doing automated provisioning of new servers and am trying to integrate puppet into this process. What I''m wondering though is what the best process for securely registering a new node is. At the moment the first time puppet is run I have to then accept the certificate on the puppetmaster and then run puppet again. What I would like to do is accept the

What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Had anyone got an idea of there exists an LDAP backend for SIP and IAX? I've read that there is a patch for LDAP realtime, but I hadn't seen any type of relevant configuration information. Any information on the above would be highly appreciated. Regards, Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have kernel 2.4.16 with ext3 compiled (not as a module), with redhat 7.1, and downloaded lilo 22.1 from util-linux 2.11n but still, on init the root fs is mounted initially as ext2 as read-only, afterwards it is unmounted and mounted again as read-write ext3. this causes problems upon unclean shutdown where I have to fsck it just like ext2. does anyone know this problem ?

I have received your answer regarding the intention of developing audio mix algorithm based on Speex (a while ago...). You've mentioned two different scenarios. VOIP applications that can manage with high bit-rates can truly mix the audio just once at the receiving client. Today most sound cards can mix the decoded streams automatically (no need for mix software). The real problem is

Anyone has an idea what happend to voip-info? it stopped working about 24 hours ago. Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/23cdc0f6/attachment.htm