similar to: cdr_addon_mysql.so did not register itselfduringload

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

Did you install the MySQL libraries? Debian's command is - apt-get install libmysqlclient15-dev Andy -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthias Urlichs Sent: 27 November 2008 16:05 To: asterisk-users at lists.digium.com Subject: [SPAM] - Re: [asterisk-users] FW: cdr_addon_mysql.so did

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

Well the addons from 1.4 are installed. This original Asterisk 1.2.x box was created by my predecessor and he had the cdr_addon_mysql.so and res_config_mysql.so files on a server that we copied to any new installation. I'm not sure where he got these files. As far as I can tell shouldn't the install of the addons create these files? If not where do I get them from? I've done a search

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 was able to get it to work with 2 extensions. One for the "host" and one for the "participants" Not the best way to set it up but it works. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, December 13, 2006 8:06 AM To: Asterisk Users

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

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

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

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

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 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"

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

Do you mean the peristence of connecting a specific phone to a specific server? If so, then it's relatively easy. The ldirectord has a persistence setting that does that. If I'm misunderstanding you, then could you explain further what you mean? Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

This is mostly in a traditional pbx-like setup. That is, these are individual remote offices of a larger corporation each with their own cluster (or clusters, in the case of one site). So there is no NAT, and it is an Asterisk-only solution (at least insofar as telephony software is concerned). Regards, - Brad _____ From: asterisk-users-bounces@lists.digium.com on behalf of David Thomas

I understand what you're saying now. While I have absolutely no proof of this, I have to believe that it's something they've solved. I've got several production systems (since early December of last year) using the type of cluster that I'm talking about, and I've yet to hear of any issues that could be related to this. I also did extensive testing both in the lab and at

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Having some trouble building cdr_addon_mysql. I've installed mysql and mysql-dev and all related rpms on a RH9 box. The box is up and running *. I've checked out the asterisk-addons but at "make install", I receive the following error: ./mkdep -fPIC -I../asterisk -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -I/usr/include/mysql -c -o cdr_addon_mysql.o

Thanks for the tip, but adding the CFLAGS directive doesn't work either, same error message. I'll try to have a look in -dev, but if anyone comes up with a solution, a reply would be appreciated. -Manuel -----Messaggio originale----- Da: Luckcuck Nick-LCKN001 [mailto:LCKN001@motorola.com] Inviato: lunedì, 21. giugno 2004 13:52 A: asterisk-users@lists.digium.com Oggetto: RE: