similar to: Asterisk 1.8 with web-meetme crash

Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881-conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881-conf-20110216-130321.wav

Hey Guy! I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten =>

Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e1f59a92/attachment.htm>

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3.

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

Dear all I have asterisk and now i want to configure meetme confranceing but problem is when i dial confrance number i got message conf number is invalide and i got this error /dev/zap/psudo device not found what is this ??? --------------------------------- Need a vacation? Get great deals to amazing places on Yahoo! Travel. -------------- next part --------------

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 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:

Hi everybody I have a problem and want to know if anyone has already seen it before : I try to use web-meetme.3.1.0 and follow these instructions http://sourceforge.net/docman/display_doc.php?docid=48924&group_id=164788 1) when i do "make" command in cbmysql folder, errors happened ********************* cc -pipe -I/usr/include/mysql -L/usr/lib/mysql -fPIC -I/usr/src/asterisk

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

I have a web page to do the usual meetme admin stuff -- mute, kick, etc. Now, the client is asking if they can listen to the meetme -- click and audio comes out the computer speakers. How can this be implemented? Is this a use case for WebRTC? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

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

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk

Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI> dahdi <tab tab> No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI> root at campbx1:/etc/wanpipe# wanrouter hwprobe ------------------------------- | Wanpipe Hardware

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

Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -------------- next part -------------- An HTML attachment was

Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back "conf-onlyperson". This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). The box is *

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",