similar to: No data when recording a Meetme conference with Monitor

Hi folks, My stack of trusty FreeBSD servers always seems to be growing, and it's getting to the point where the daily and security output mail is too much to make good use of. I'm looking for suggestions for log monitoring and aggregation tools, especially from a monitoring-for-security perspective. If I had to imagine an ideal system, it would be a central server that securely

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hello. Hopefully I didn't miss this in the archives, so here goes. I've been running World of Warcraft in the CVS version of wine. The game runs perfectly as far as I'm concerned. However, if I run the game at a resolution lower than the desktop resolution it's not quite right. It switches to a view port of the specified size rather than resizing the whole desktop. this means I

Hi, Has anyone connected * to a Colt E1 line in Europe? If so could you send me the zaptel.conf and zapata.conf. Thanks, Aaron __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Hi all, I just recently began using HTB to try and manage bandwidth for my network. This is the script I''m using: /sbin/iptables -t mangle -A FORWARD -o eth1 -s ! 192.168.244.2 -j MARK --set-mark 53 tc qdisc add dev eth1 root handle 1: htb default 20 tc class add dev eth1 parent 1: classid 1:1 htb rate 100mbit burst 131072k quantum 59000 tc class add dev eth1 parent 1:1 classid 1:10

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

I am looking for a good Asterisk GUI to manage my server. Any Suggestions? Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/96bf8fea/attachment.htm

Folks, I've been running asterisk successfully using the extensions.conf and voicemail.conf. Now that I've got asterisk happily looking up MySQL tables for the VM configuration, I decided to try out the contributed script /usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl I edited the script so that its output goes to a separate extensions_from_mysql.conf file. The

OK. I now have call recording working for both incoming and outgoing calls. Now I want to make those wavs into mp3. I could launch a script from cron that checks for new wavs and converts them. But that wouldn't be so elegant. Launching it from * on hangup would be nicer. How is it done? [outgoing] exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten =>

Full_Name: susscorfa Version: 2.7.1 OS: ubuntu Submission from: (NULL) (129.125.177.31) Incorrect implementation of the grouping variable in the function glmer crashes R a small example: require(lme4); a<-data.frame(b=rpois(1000,10), c=gl(20,50), d=rnorm(1000,3), e=rnorm(1000,5), f=rnorm(1000,2)+5); glmer(b~d+f|c+(e), family=poisson, data=a) It crashes R on debian linux (2 independant

I have a voicemail application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5)

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

I'm having some issues with Background() and Read() commands. See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'. All works fine. hestia*CLI> -- Executing Answer("SIP/2944093-3366", "") in new stack -- Executing Wait("SIP/2944093-3366", "1") in new stack --

Hi all, I want to setup a meetme application in the following maner: One operator is connected to a room. The operator hears and can talk to all the participants, but one participant can only hear/talk to the operator, not others. The operator is using one phone. To be more explicit, this means that every new person etering the room has a one2one conversation with the operator only, and the

Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks.

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean

.. and from a newbie no less :-) I have configured my BT101, and hooked it up to my * box. All is well. I have entered the following in externsions.conf, and this bit works: exten => 613,1,Answer exten => 613,2,Playback(demo-echotest) exten => 613,3,Echo exten => 613,4,Hangup If I pick up the BT101, and dial 613, sure enough I get the echo test.. All good. I have a TDM400 Card

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"