similar to: user number in conference

Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002", "CHANNEL(language)=fr") in new stack -- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002", "") in new stack -- Executing

Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the room. It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ? can anyone help me and maybe someone has also the problem as i and have an solution. I use: asterisk-1.6.2.7

All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way?

Hi Guys, in next week i plan to upgrade my Asterisk Server. To buy the optimal Hardware i have a question. What is better more cores (eg. 2x quadcore) or more CPU speed for a server that handle a lot of of Meetme Concerences with hundreds of concurrent G711 alaw Channels (no transcoding) ? in my opinion, more cores are better, because Asterisk ist multithreded and each channel has a good

Hi, I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the user and call AgentLogin. But after that when I call AgentRequest I keep getting Agent '1234' is busy. If I put a delay of 5 second or more before calling AgentRequest then it works most of the times. Here's my dialplan: [login] exten => s,1,Background(thank-you-for-calling) same =>

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

How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100414/ba26a927/attachment.htm

I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According

hi, I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. Thanks for your help, Emrah

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

Hi Stream-Freaks, I have an online radio, and very often the stream is broken, in the logfile is the error message [17/Jan/2003:16:10:59] [2877:Source Thread] Kicking client 2877 [195.145.211.50] [Too many errors (client not receiving data fast enough)] [listener], connected for 13 minutes and 55 seconds, 5196648 bytes transfered. 0 clients connected I don't know how. The Client has a 2Mbit

Has any Sun Solaris samba admin ever attempted to mount to a WindowsNT server? Unfortunately after several attempts and some elaborate syntax I can not get this to work. I do not see that Sun supports smbfs as Linux does. If anyone has any ideas or tricks please drop me a line. Thanks, Steve Knoll Intersil Corporation PAT Systems Group * (email) < mailto:sknoll01@intersil.com> *

_History_ I have 100% success in Xen VGA Passthrough with Xen 3.5-unstable with my first Palit NVIDIA Geforce 8400 GS VGA card in 2009, which is 3 years ago. Then my first Palit NVIDIA Geforce 8400 GS VGA card overheated and malfunctioned. So I bought a 2nd Palit NVIDIA Geforce 8400 GS VGA card several months ago in March 2012 for S$44, to replace the first card. At that point in time, Xen

_History_ I have 100% success in Xen VGA Passthrough with Xen 3.5-unstable with my first Palit NVIDIA Geforce 8400 GS VGA card in 2009, which is 3 years ago. Then my first Palit NVIDIA Geforce 8400 GS VGA card overheated and malfunctioned. So I bought a 2nd Palit NVIDIA Geforce 8400 GS VGA card several months ago in March 2012 for S$44, to replace the first card. At that point in time, Xen

I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten => _XXXXXXXXXX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten => _XXXXXXXXXX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can

Hello Technical Guys, I have follwing problem and I hope someone has an idea. My stream Server is icecast and I use shout to stream the mjusic to the server. Sometimes the listeners where kickted, and I don't know what it is. the logfile icecast.log shows me following entry [12/Jun/2002:05:23:51] [1158:Source Thread] Kicking client 1284 [pd9556401.dip.t-dialin.net] [Too many errors (client

> > > > If it is that then either the solaris send window or the > client receive > > window is too small. > > > > Or, we could go for a vastly less complex explanation - shout > is known to not > send at the correct rate. Don't use it. > > Mike <p>Hi all, I understand, what for a livestreamer can I take ? any references ? A network

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx

Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing

Hi, I know this question may have been asked before (although the archives don't seem to suggest it), but has anyone had any problems with Asterisk accepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf =>