similar to: Anyone tested the new Sony Ericsson P1 phones..

Hi Experts, I would like to present time series data in meaningful way in building some graphics. I've tried: (1) plot(ts(mbaye3)) and (2) plot(ts(mbaye3), start=1990) But I always get this error-message: Fehler [error] in plotts(x = x, y = y, plot.type = plot.type, xy.labels = xy.labels, : cannot plot more than 10 series as "multiple" my data: mbaye3

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

In looking at the safe_asterisk script, it would appear that it is encouraging the running of the Asterisk application as root user. My natural inclination is to run it as a non-privileged user. What is recommendation? +++++++++++++ This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415

Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open?

Hello to all I need send a data to sofphones screen when I use a Dial () . Thanks a lot Regards Andres Gomez -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070430/d26033b2/attachment.htm

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

HI I have 3 Linksys SIP901 IP phones I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd I'm looking to connect this phones together and to make calls between them Not from outside of my lan I don't know how to configure asterisknow beta Can somebody help I'm doing this in my house to connect rooms With respect Ardit Saliu

Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can

Thanks to all that have helped me so far. I have made a lot of progress. I am able to make prilib and zaptel. Now to Asterisk... After installing the kernel source, I have: # cd /usr/src/linux # make cloneconfig # make prepare-all Then I have run ./configure in the asterisk-1.4.4 directory. I have: # make clean # make Which goes through a number of compiles and then ends

Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced

Can one server (like AMD 6000+ X2) with 2 GIG ram running asterisk 1.4 handle having 2100 wireless phones connected. All phones will not be talking at the same time only a couple will be. There may be 1 T1 card in the box. Will this work? If not how does one handle this situation. Thanks, Jerry

Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with "iffy" DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS

Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi:

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

Hi all, * i am using widows based asterisk pbx(AstWin) which i have down loaded from www.asteriskwin32.com . we r using x-lite as a soft phone now .we have created 10 sip users in sip.conf and configured extensions.conf too. all of us could make calls through asterisk. we made 10 calls at the same time through one peer. now i wanted to transfer the call from one user to another. how can i do

Hi all, Does ENUMLOOKUP can query multiple DNS servers without having to replicate the same code in which the only thing replaced is the server? If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to find the list of DNS servers in order of preference to be queried, but, I pretend to use something like this: ${ENUMLOOKUP(+${ARG1:2:},sip,c) which does not seam to care about

Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [root@localhost /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [root@localhost /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the

Hello, we are running a Asterisk (1.2) installation with about 80 snom phones (300,320,360). Now have the demand for a special manager - assistant setup for a few extensions. Since Shared Line Appearance is not available in 1.2 I?m wondering how to realize this... What we need is that the manager can decide whether he wants to get calls or not. If not he must have the possibility to redirect

I have all my SIP users in a realtime database. I would like to use MySQL command to query the database and use the results from the query to page all the phones found in the query. The results from the MySQL query will be multiple rows of extension: Something like: mysql> Select extension from sip where extension like '6%' 6001 6002 6003 ex.... I need to put all the results into a