similar to: Syntax Help on a Bash Script

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I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

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So no one else is having issues with MySQL and 1.4? I'm the only one? -----Original Message----- From: Savoy, Kevin - Williston, ND Sent: Wednesday, December 27, 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Well the addons from 1.4 are installed. This original Asterisk

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Ok so something is missing. I get the below for those two lines. checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... no I even installed the mysql-devel as Bradley Watkins suggested and still it says no. What do I need to make that say yes? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it actually posted. The below worked for normal transfers. Now here is another situation. When we try to transfer a call directly to voicemail it plays the voicemail message but we can't transfer the call. The only way I could get it to work was to do a conference and then drop out of that conference. My

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

I am having an issue with 1.4 where we can't successfully transfer a call directly to a voicemail box. We hit "Transfer" on the phone and dial the mailbox number we want to send it to, My dial plan for this is: exten=>_*40XX,n,Voicemail(${EXTEN:1},u) The voicemail system picks up and starts to play its message and at this point. We should then hit "Transfer"

Hi all, I am a biggener in R-Project I got one problem with applying *where condition* like if 2 tables like table1: empid name dep 101 kiran solutions 102 ram testing 103 pavan database table2: empid month sal 101 Dec 9500 102 Dec 9800 103 Dec 8500 in first table i have to take *empid* with

I have a string like this st <- "SELECT COUNT(empid), COUNT(mgrid), COUNT(empname), COUNT(salary), FROM Employees" How can I remove the last comma before the FROM statement? -J

I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I

I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk -vvvv I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]:

I have discovered an issue on my system after upgrading from 1.2.13 to 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I have confirmed this on multiple phones. When the called person answers and tries to transfer the call to another extension, the call successfully transfers, however the person answering the transfer cannot hear the person that called in, the caller. My

This worked. Great and thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Carlos Chavez Sent: Wednesday, February 07, 2007 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly On Wed, 2007-02-07 at 14:12

I need some help with some statements..... #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); my $callerid = $input{'callerid'}; if ($optemp != 1) { my $empid = $AGI->get_data('employee',-1,5); $AGI->stream_file(entered); $AGI->say_digits($empid); my $optemp =

Hi All. In my site I have situation that I always want to execute some piece of code when AJAX call happen. I am using function onCreate and onComplete from tutorial:http://www.sergiopereira.com/articles/prototype.js.html but I don''k how to manipulate Ajax responseText in this function. Does anyone know how to do it? Gregor ---------------------------------------------------- Zagraj o

We are looking at using Asterisk as a call recording server for an Avaya VoIP S8700 system in a multi-site VoIP Call Center. All calls will be coming in to one location and sent out via VoIP to other call centers. What kind of specs should we be looking at purchasing for our Asterisk server to be record up 200-300 calls simultaneously? Linux runs in 64 bit architecture, but does Asterisk

I am caught in a mental trap. Why isn't the between groups variance estimated (0.0038) to be around the value with which I generated the data (0.0002)? Thanks Toby set.seed(76589437887) fph = 0.4 Sigh = sqrt(0.0002) Sigi = sqrt(0.04) ci = 1 fpi = matrix(,7200,3) for (i in 1:90) { fph = rnorm(1, fph, Sigh) for (k in 1:80) { fpi[ci,1:3] = matrix(c(i, k, rnorm(1, fph, Sigi)),1) ci

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif