similar to: External sip phones register with the servers IP...

Hello I am using sem to look at the direct effect of one variable on another but i am uncertain if i am progressing correctly. An example: covar1<-? matrix(c(0.4,-0.2,3,-0.2 , 0.3,-2 , 3 ,-2 , 60), nrow=3,byrow=T) rownames(covar1)<-colnames(covar1)<-c("endo","exo","med") path1<-matrix(c(? ? "exo -> endo",? "g1", NA,

Hi, I am brand new to R and not familiar with the language, though I have been reading the manuals and making some slow going progress. I am working with some source code from a Global Vector Auto -Regressive program written by Ranier Puhr from the R-forge group. I need help interpreting the processes of the following code. I am going to post in parts since it's pretty long: GVAR

dear: when I read a Excel file(exp-11),The R project give me a error ,Just like this: ??'datafile' > write.foreign("exp-11.xls") ???list(df = df, datafile = datafile, codefile = codefile) : ??'datafile' > write.foreign("exp-11.xls") ???list(df = df, datafile = datafile, codefile = codefile) : ??'datafile' >

Hi, it's my first time to write a loop with R for my homework. This loop is part of the function. I wanna assign values for hll according to panel [ii,1]=pp. I didn't get any error message in this part. but then when I further calculate another stuff with hll, the function can't return. I think it must be some problem in my loop. Probably something stupid or easy. But I tried to look

I am fitting a reduced form VAR model using VAR in the vars library. I have several endogenous variables, and two exogenous variables. I would like to explore the effects of a shock to one of the exogenous variables on one of the endogenous variables. Using irf in the vars library only calculates the irf for the endogenous variables, this is obviously by design, is there some theoretical

Hello, we have installed non-repository based software on our centos servers (e.g. vmware tools). Each time we do 'yum update' we have to run several scripts to check if these software works after update. Sometimes we forget ... It it possible to configure any sort of triggers which will run automatically after 'yum update' ? I'd like to avoid writing shell wrappers for yum

I am using the latest xlite phone to connect to the latest version of asterisk (20040727). When I try to make a call the xlite phone tells me "Call not approved". I used the configuration options that were listed on the wiki. The context in the sip.conf file is "from-sip". I have a matching context listed in the extensions.conf file. The phone is able to register

Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build

Hi, I am using reticulate and a virtual environment (not conda) to run Python scripts from RStudio. However, when I try to use my own (existing) virtual environment, reticulate does not use it. If I run my scripts, the installed modules (e.g., py_install("pandas", "mmstat4.hu.data")) are not found. I believe this happens because reticulate is using r-reticulate instead of

Have you tried https://rstudio.github.io/reticulate/ ? Generally speaking, complex nonstandard package specific questions such as yours rarely get a reply here -- there are 20,000+ packages (and counting) after all! As reticulate was created by and integrated with RStudio/Posit, I would think their site and help resources might be a better venue. Of course, if you don't use RStudio, you may

I have been using 11.4.0 for some time. All was fine. I downloaded 11.5, extracted, run ./configure, make, make install. I got a message about res_rtp_asterisk.so was not compiled in the 11.5 Sure enough I have rss_rtp_asterisk.c but not .o file and no .so file. I then looked in the config.log and nothing is in there about res_rtp_asterisk What's up? jerry -------------- next part

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make

Hi After upgrading from 1.4 to 11.4.0, I am able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx at Accueil_HNO:2] BackGround("SIP/SIP000005-0000000c", "Fermeture") in new stack [Jun 16 07:35:06] WARNING[7634][C-00000006]: file.c:701 ast_openstream_full: File Fermeture does

Hi? As I seen on the Asterisk web site , there is packages called ,? AsteriskLatest Version - 11.4.0 asterisk-11-current.tar.gz?and? asterisk-1.8-current.tar.gz May I now which one is the most suitable for a production environment ? Thanks in advance Luke -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I need to rebuild a system which has 4 BRI ports and is connected to Point-to-multiPoint lines, in a country where telco often "drop lines for energy savings". I'm planning to use latest 11.4.0 asterisk version along with dahdi and libpri (no misdn). Which version are recommended for Dahdi and Libpri ? My main requirements are, beside having calls coming in and out: - keep a

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port I get a motorboating sound or warble - or - just not clear audio. When I switch that to ALSA direct it sounds just fine. What might be happening with pulse audio that it does not sound clear??? asound.conf below. Thanks, Jerry more /etc/asound.conf # # Place your global alsa-lib configuration here... # @hooks [ {

Hi We want optimize my extensions file conf on asterisk 11.4.0 : We have a big quantity of extensions, all are same "design": ; Destination: Gambia Type: Fixe exten => _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) exten => _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exten => _00220X.,3,Set(CALLERID(all)=${NUMID}) exten =>

dear all, This is a probably a silly question. If I type > grep("x",c("a.x" ,"b.x","a.xx"),value=TRUE) [1] "a.x" "b.x" "a.xx" Instead, I would like to obtain only "a.x" "b.x" How is it possible to get this result with grep()? many thanks for your attention, best, vito --

Hi all, We're using samba, and we'd tried it with multiple Windows versions: 98, ME, 2000 and XP, in that order. However, when we migrated from 2000 to XP, a problem apeared that was not present in any other Windows version. Everthing works fine, excepting that when I am using the samba shares (browsing, opening files, etc...) from time to time I get a ~30 second lockup. During that time,