similar to: ASTCC: perl / mysql or me???

I try to track down an error that causes that Astcc just reports the time, but not the costs. I could narrow the problem down into this sub routine: sub calccost() { my ($adjconn, $adjcost, $answeredtime, $increment) = @_; eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment }; my $cost; print STDERR "Adjusted time is $adjtime, cost is $adjcost with

Is there a reason that the bottom axis changes color when POSIX data is used in plot function? For example: > timedata <- c("2/3/2003","3/4/2003","5/4/2003") > timedata2 <- strptime(timedata,format="%m/%d/%Y") > numdata <- c(2,3,4) > plot(as.POSIXct(timedata2),numdata,col="red",type="o") As compared to: >

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR

Hi, my name is Julio, I'm trying to run a script, specifically with the siar package on R (I have attached the script) but the program collapses on during the run. The error message says: *** caught segfault *** address 0xb31941cc, cause 'memory not mapped' Traceback: 1: .C("siarmcmcv4", as.integer(numdata), as.integer(numsources), as.integer(numiso),

Hello Lady or Sir, I want to do model selection with 'step' function. But there is a trouble to me. For example, there are 5 independent variables, X0,X1,X2,X3 and X4; and 1 dependent variable y. I want to run stepwise regression y ~ X0 + X1 + X2 + X3 + X4, and I want to keep X0 always in the model. Every time when I run command 'step' with 'keep', error information

Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc.

Hi, all Sorry that forgot add mail subject last one. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database

Hi, all Sorry for null subject last mail. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database

I am currently trying to write a program that minimises the amount of work required for “auditable” qPCR data. At the moment I am using an Excel (.csv) spreadsheet as source data that has been transposed to the column format required for R to read. Unfortunately, this means I have* *to manually confirm the whole data set prior to doing any analysis, which is taking a considerable amount of time!

Hello everybody, I want to use the nnet library and my problem is that the algorithm seems to accept only class as target when the neural is fiiting. So the output when I use predict.nnet is also a class. Is it possible to have numeric variable as target ? If yes, what is the syntax ? Thank you. Cordialement, R?gis CHARIGNON

Hi! I trying to configure * in a way, that it uses a different CLIP (Caller-Id in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far always the main (1st) number of the number-block is sent to the ISDN. I have a E100P from Digium and use the zapata stuff (chan_zap). All SIP calls are coming through an SER. One idea I had in mind is to assign userid's in SIP, that match

I have some troubles with ASTCC. TOOOOO often the "in-use" flag remains set. I would like to find a solution, where astcc.agi checks automatically if THIS user is in a call rather than to check the flag. If that is not possible, I would like to have an extension to dial to, and it will after hang up, reset the flag! The in-use flag remains set, if the caller hang up before the

hello, I am beginning to asterisk. I have a sip trunk access to operator and VPN access with operator. i booked 10 sda numbers. IP adress asterisk : 192.168.600.1 IP adress operator : 192.168.700.50 i can ping on 192.168.700.50 # cat sip.conf [general] context=default srvlookup=yes port = 5060 disallow=all allow=gsm allow=alaw allow=ulaw [1000] username=1000 type=friend qualify=yes

Hi, I am out of town and will get back to you on the 13th of July. Leo >>> "r-help at stat.math.ethz.ch" 06/27/03 00:32 >>> Send R-help mailing list submissions to r-help at stat.math.ethz.ch To subscribe or unsubscribe via the World Wide Web, visit https://www.stat.math.ethz.ch/mailman/listinfo/r-help or, via email, send a message with subject or body

Hi, alhtough i googled for details concerning ASTCC i did not found an aswer to the following: Should i put in my extensions.conf the configuration of the astcc? I ask this because as i see it, in the end of the extensions.conf there is an include statement : #include /var/lib/astcc/astcc-exten.conf Should the config been done in the astcc-exten.conf file or the initial extensions.conf

Nicolas, Thank you for your response. I had tried that before and it didn't work. I am trying to look up the route for a dialed number, so its a full E.164 number. Please see my query below when I try to look up the route for a USA number; mysql> SELECT * FROM routes WHERE "^13237309880" RLIKE pattern ORDER BY LENGTH(pattern) DESC;

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

I have not used astcc with pin codes so far, since I set-up the phone number as card number. Some of my users want now to dial in to the system and than use their card, which is their phone number. For that I would need a way of authentication, like a pin. I want to use something like: What is your card number: <user keys in the number> Enter your pin: <user enter a long pin>

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Does anyone have a mysql query that will compare a number from the asterisk cdr to a table of international country+city codes to determine the closest match? The two fields are; 1. Asterisk mysql cdr 'dst' field - sample record value '011441316551212' 2. rate table data like this DialPattern 011447977 011447979 011447980 011447981 011447984 011447985 011447986