similar to: Call Center Phone with Auto Answer

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

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

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Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum

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

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

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"

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

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

Hi All, I'm brand new at doing anything linux and would like feedback on this script I'm trying to understand from an example I'm working on.. Oh, running Centos 5.6 Anyhow, I run this bash script: #!/bin/bash # send data to the table in the MySQL database MYSQL='which mysql' if [ $# -ne 4 ] then echo "Usage: mtest4 empid lastname firstname salary" else

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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

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

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

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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]:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 ==== Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Olivier 334 Philipp Kempgen 251 Gordon Henderson 242 Atis Lezdins 239 Jay R. Ashworth 230 Doug Lytle 207

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