similar to: Meetme Participant Announcement

Anyone seen this? If not I guess I'll have to post it as a bug. Extensions.conf has this: exten => 123,1,Meetme(|dMic|) I dial 123, and enter my conference number. Asterisk asks me to enter my name. At this point I hang up. If I type at the Asterisk console 'meetme list 12345' it shows that I am a participant in the conference evenhough I hung up. If I dial 123 again and this

Hi guys, Having a bit of a problem trying to get conference bridges working. In my meetme.conf file I have the following line [rooms] conf => 6000 In my extensions.conf file I have: exten => 1000,1,MeetMe,6000 My problem is that when I dial into extension 1000 it is telling me "this is not a valid conference number". Can anybody telling me what I'm doing wrong here?

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

I'm trying to record a Meetme conference to disk, but the Monitor application doesn't seem to play nicely with Meetme. In extensions.conf, I have this: exten => 1000,1,Answer exten => 1000,2,Monitor exten => 1000,3,Meetme This starts up the monitoring OK, and it records the prompts that Meetme gives, but as soon as the user enters the conference, the -out WAV file stops

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

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, 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

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

Dollars to donuts it is related to these two posts, but no one seems to know where or why it happens - this issue doesn't seem to be related to one specific piece of hardware: Post 1) ********************************************************* Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while

Hi, I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? thanks. Eyal Mahalal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in again. When dialing in again the delay seems to go. It seems to me as though as soon as the server registers

I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference "room", ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

Hello everybody, I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE mode, the other one in NT mode. Whenever I call into an empty MeetMe conference room on one of the NT ports or via SIP and hang up the call during the "you are currently the only participant in this conference"

hello is it possible to lock a conference IF no admin are connected ? or how to do to have a conference offline? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon

Is it possible to record a meetme conference? What channel would you monitor, is there a main channel that all audio goes too? If so, is it possible to use the ast_monitor (iirc) to record that channel? Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 *