search for: bergamini

I need a table showing even the zero counts. x<-c(1,1,1,2,2,2,5,5,5) table(x) x 1 2 5 3 3 3 How can I get this: x 1 2 3 4 5 3 3 0 0 3 Thanks, [[alternative HTML version deleted]]

Hi all, sorry bothering again. I am still stuck in compiling asterisk. Learning (or trying to) from the first problem and first hint, when I got this error: gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in

Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). ***** A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium,

Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out

I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical

asterisk-users-request@lists.digium.com is believed to have said: >I don't mean to be rude to everyone who responded to this question, but >I think that everyone is answering the wrong question. The point is that >the message waiting indicator doesn't light up, at all, ever. All that >happens when messages are waiting is that the display blinks and the >phone gives a

asterisk-users-request@lists.digium.com is believed to have said: >I am using slackware 10.1 (kernel 2.4.29) and I am getting the following >when I issue gcc -v Dimitris, while I never compiled chan_capi I thought you would need a 2.6 kernel to use it. HTH Aldo

asterisk-users-request@lists.digium.com is believed to have said: > >and watch linus himself rant about how this is incorrect to do (yet all >the distros do it) :P > Well, this is reassuring for a newbie like me. Even the pros (as anybody building a distro ought to be, and most of the times, really is) can do obvious errors... Aldo

asterisk-users-request@lists.digium.com is believed to have said: > >The active cards do the ISDN protocol stuff on board, so the host CPU/driver >does not need to do that -> better performance, less interrupts. >The AVM cards do not have such DSPs on board, so no echo-cancel. >But the Eicon DIVA Server cards do. They do analog Fax/Modem, echo-cancel, >DTMF-detection, voice

asterisk-users-request@lists.digium.com is believed to have said: >chan_capi does not set the NT-mode. Your cards driver need to do that. >E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl >or set NT-mode in the config wizard. >chan_capi does not (need) to know anything about what protocol the card is >doing. CAPI is independent here. Ok.

Hi all, I have noticed something I can't solve regarding Asterisk (latest 1.6.0.x). My server is set at the GMT+2 timezone. The clock is ok (I can get the correct time at the terminal). But today I got a call at a time where Asterisk should have gone 'off business hours'. All log times are wrong by exactly 2 hours. As if Asterisk would just sit on GMT, ignoring the GMT+2

asterisk-users-request@lists.digium.com is believed to have said: >Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a >protocol for the phone to talk to any Bluetooth headset, no matter who made >it. This protocol would have to include something to allow voice to pass >from the phone to the headset and vice versa. It might also include >something for

asterisk-users-request@lists.digium.com is believed to have said: >Budgetone and MWI > >The message button can be programmed to dial an extension that checks >voicemail >exten => 160,1,Voicemailmain(${CALLERIDNUM}) > Thanks, this is what I was thinking about. Still, how do you get the BT to dial 160? In my Asterisk setting I have the same mailbox numbers reused for the

asterisk-users-request@lists.digium.com is believed to have said: >> While we are at the subject another couple of simple related question. >> >> Are HFC-S cards active? I got one for a very low price, so that I >> imagine it will be NOT the case... > >No, these cards are passive. The protocol handling must be done by >the hosts CPU / driver. OK; price

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242

...rom: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Cotton Sent: 22 July 2005 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: zaptel make problems On Fri, 2005-07-22 at 07:59 +0200, Aldo Bergamini wrote: > asterisk-users-request@lists.digium.com is believed to have said: > > > > >and watch linus himself rant about how this is incorrect to do (yet > >all the distros do it) :P > > > > Well, this is reassuring for a newbie like me. > > Even the pr...

asterisk-users-request@lists.digium.com is believed to have said: >Hi, > >Sorry for being very late on this thread but i am trying to make a >decision on which one to go for. Options are > >1. Dock n Talk offered by Voxilla (USD139) >2. GSM Gateway by CyberTelecom (GBP60) > >I'm having a TDM400P with 1 FXO & FXS. > >I'm interested in implementing DISA

Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound-capi-local] exten => _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn CAPI/${CALLERIDNUM}) exten =>