similar to: probleme with web-meetme.3.1.0

I am hoping maybe some of you have come across these before in your experience with web meetme. Below are the messages im receiveing when I load the web meetme home page. Notice: Undefined variable: s in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 9 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 12 Notice:

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten =>

I am experiencing the same problem. Fresh install. Bill -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ma Zhiyong Sent: Tuesday, January 30, 2007 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] web-meetme cbmysql not registered HI, today I download

I tried this with a fresh tree, so I know it works. Based on your last message, I think you may have missed step 1. My needs are for both CBMysql and MeetMe2, so I bundled the patch file. If you are only interested in one, then delete the sections of the patch that are not needed. 1. Download the source for MeetMe2.c from http://www.areski.net/asterisk-meetme/about.php?s=0 and put it in

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

http://bugzilla.mindrot.org/show_bug.cgi?id=88 Summary: scp for OpenSSH v3.0.2p1 fails to commercial SSH v3.1.0 server (new -t option to SSH's scp) Product: Portable OpenSSH Version: 3.0.2p1 Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: scp

When trying to boot the OpenSolaris b75a dvd in a pv domU (using the linux phy:/dev/cdrom device as the source for the boot media), running on an Ubuntu 7.10 x86_64 dom0 box with xen 3.1.0 hypervisor, using the following config file... ======================================== # # Hostname # name = ''solaris-install2'' # # Kernel + memory size # kernel =

Once again, I know that a true MeetMe2 is in the works, but if anyone is looking for a database conferencing setup then check this out. New diffs and PHP web interfaces at www.fitawi.com/Asterisk -New in the interface: Add a conference Modify an existing conference -New in app_cbmysql: Renamed confmysql.conf to cbmysql.conf Move conference application name to conf file Move conference

Lots of interest here for conferencing. I've probably convinced more people to start using asterisk@home for this feature than anything else. Can I input some suggestions; Need to change the rinky dink call icons (let me know if you need some better samples) Need to change the ability for a muted caller to flag they have a question and a desire to talk (I do a lot of mute all except 1 or 2

Last update before I head to Korea for an IP phone deployment New diffs and PHP web interfaces at www.fitawi.com/Asterisk -New in the interface: New fields for conference conference owner and title Delete conferences Changed date/time to listbox to reduce user input error -This will likely have to become a java or other client-side code, since I don't see a way to tie the number of

On 1/3/07, asterisk-users-request@lists.digium.com < asterisk-users-request@lists.digium.com> wrote: > > ---------- Forwarded message ---------- > From: "Dan Austin" <Dan_Austin@Phoenix.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > Date: Wed, 3 Jan 2007 10:33:02 -0800 > Subject:

I'm afraid I am at a loss. If the three files, app_cbmysql.c, app_meetme2.c and Makefile all exist in ../apps then a patch -p1 from the ../asterisk directory should work. The -p1 tells patch to ignore the first directory in the path to the file in the patch, -p2 ignores two directories. Another option is to just edit the apps-meetme-cbmysql.txt and split it into three patchs and apply

We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which

Hi, I'm trying to setup Web-Meetme 4.0 and I always get the following warning when I open the default page http://localhost/web-meetme Warning: session_start() [function.session-start]: Cannot send session cache limiter - headers already sent (output started at /var/www/web-meetme/locale.php:36) in /var/www/web-meetme/meetme_control.php on line 34 Has anyone a solution to this? Cheers

I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also installed the latest versions of mysql and php. I followed the readme file that came with the web meetme app and everything seemed to go fine up until I realised the module wasnt being loaded. When I stop asterisk and try to start it, it errors out and does not load and I get the following message: Parsing

A couple comments. I'm not a programmer, my C is passable, but my web development would have to grow by leaps and bounds to be considered poor. I pulled the Meetme2 from here: http://www.areski.net/asterisk-meetme/about.php?s=0 The app needs a minor tweak to compile against 1.0.5. I stumbled down a false path or two, so the diff shows some lines being deleted and re-added that are

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

Hello, I just checked out the latest CVS and compiled and now get the following error: [res_config_mysql.so] => (MySQL RealTime Configuration Driver) Jan 26 13:03:51 WARNING[27081]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/res_mysql.conf': Found Jan 26 13:03:51 WARNING[27081]: res_config_mysql.c:561 parse_config: MySQL

Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction ? ast_config_load ? make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? --

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong